- sr-users - kamailio.org

RE : [Serusers] SER + Radius + B2Bua
by lists＠cingerr.com 07 Feb '06

07 Feb '06

Im still seeking my head how Vonage and others can do that? does Iptel have any solution for that, if yes how much may that cost? > Depends of what you want to do... > For a real time billing for prepaid, you will need a b2bua, Ser will not give you the possibility to do that(I think). > For postpaid, Ser is enough. > > Olivier > > -----Message d'origine----- > De : Waldo Rubinstein [mailto:waldo@trianet.net] > Envoyé : mardi 7 février 2006 14:15 > À : Olivier Taylor > Cc : lists(a)cingerr.com; serusers(a)lists.iptel.org > Objet : Re: RE : RE : RE : [Serusers] SER + Radius + B2Bua > > > Would this also be a viable solution for terminating calls to other SIP gateways instead of just PSTN or would SER-only be sufficient for that? > > Thanks, > Waldo > > On Feb 7, 2006, at 7:54 AM, Olivier Taylor wrote: > >> Well we use it for pstn calls and have canreinvites=yes, then >> Asterisk is >> most of the time outside of the media path. >> The only job of this server is the billing and routing calls to pstn. Furthermore, the number of simultaneous calls Asterisk can manage depends on >> the hardware u will use, transcoding and so on. >> We had more than 50 simultaneous calls on a p4 with 2Gb ram, I >> really don't >> (yet) know what will be the limit. >> >> Olivier >> >> -----Message d'origine----- >> De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la >> part de lists(a)cingerr.com >> Envoyé : mardi 7 février 2006 13:42 >> À : serusers(a)lists.iptel.org >> Objet : Re: RE : RE : [Serusers] SER + Radius + B2Bua >> >> >> Dear Olivier, >> >> Im interesting to know how asterisk behave in heavy load? how many calls at >> a moment is your asterisk handling? >> >>> Hello again, >>> >>> Just try asterisk b2bua, it works fine for us. >>> http://developer.berlios.de/projects/b2bua/ >>> >>> The developper is very kind, subscribe to his mailing list. >>> >>> Good luck, >>> >>> Olivier >>> >>> >>> -----Message d'origine----- >>> De : serusers-bounces(a)lists.iptel.org [mailto:serusers- >>> bounces(a)iptel.org] De >>> la part de lists(a)cingerr.com Envoyé : mardi 7 février 2006 11:54 À : serusers(a)lists.iptel.org >>> Objet : Re: RE : [Serusers] SER + Radius + B2Bua >>> >>> >>> Hello Olivier, >>> >>> I have about two week spending lot of time trying to make it work but no luck, so thank you for saving my time. >>> >>> Is there any known way to make prepaid/postpaid billing with SER? >>> >>> >>> Best Regards >>> Hekuran, >>> >>> >>> >>>> Just a question, do you use Vovida b2bua? >>>> >>>> If yes, forget it, development has been stoped 2 years ago and it's very incomplete. For example, Vovida b2bua is unable to manage re-invites. >>>> >>>> Just to avoid wasted time for you >>>> >>>> Olivier >>>> >>>> -----Message d'origine----- >>>> De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de lists(a)cingerr.com Envoyé : mardi 7 février 2006 10:42 À >>>> : serusers(a)lists.iptel.org Objet : [Serusers] SER + Radius + B2Bua >>>> >>>> >>>> Hello, >>>> >>>> Im trying to make a combination of radius and b2bua to make some prepaid/postpaid billing. Till now I have managed to make SER talk to >>>> b2bua and b2bua talk to Radius, but Im not clear yet where does ser or radius stores the call detail records (radacct table is always empty). >>>> >>>> Also my question about prepaid: is there any database schema that provides tables where I store user credit or so (as it says that b2bua have bundled billing system) or should it be done using radius attributes? If so are there any extra attributes that I have to add in dictionary? >>>> >>>> Question about ser: When a user registers with radius it >>>> registers as >>>> username@domain (100(a)some-domain.com) but when I try to call, SER send the call to b2bua using only username ([100]) so the user can not be found. is there any way to register the user only by >>>> username? >>>> >>>> Best Regards >>>> Hekuran >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] SER + Radius + B2Bua
by lists＠cingerr.com 07 Feb '06

07 Feb '06

Hello, Im trying to make a combination of radius and b2bua to make some prepaid/postpaid billing. Till now I have managed to make SER talk to b2bua and b2bua talk to Radius, but Im not clear yet where does ser or radius stores the call detail records (radacct table is always empty). Also my question about prepaid: is there any database schema that provides tables where I store user credit or so (as it says that b2bua have bundled billing system) or should it be done using radius attributes? If so are there any extra attributes that I have to add in dictionary? Question about ser: When a user registers with radius it registers as username@domain (100(a)some-domain.com) but when I try to call, SER send the call to b2bua using only username ([100]) so the user can not be found. is there any way to register the user only by username? Best Regards Hekuran

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[Users] received-Parameter in usrloc
by Andreas Granig 07 Feb '06

07 Feb '06

Hi, Just a question regarding the received-Parameter in usrloc: Is it ALWAYS in the format "sip:<ip>:<port>" or may there also be variations like "sips:<ip>:<port>"? What about TCP-connections? Could there be a transport identifier? Thanks, Andy

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[Users] client could not get server certificate error
by Pjothi 07 Feb '06

07 Feb '06

Dear all, wen a sip client(minisip) tries to connect with OpenSER with TLS enabled on both sides, I get the following error ipMessageTransport: sendMessage: creating new socket Creating new SSL_CTX SSL connect: Error in system call. Could not get server certificate SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: exception thrown! regarding the certificates, using the scripts given in OpenSER, I created a rootCA and with the same rootCA I created the following for server and client Server: server-cert server-privkey server-calist Client: client-cert client-privkey client-calist and loaded the sever certs in the server config file and the client certs in the client config file..... wat am I missing here, why is it not able to get the server certificate ? I am posting the openser.cfg here. kindly guide me thanks and regards, Pjothi ____________________________________________________________ openser.cfg # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support disable_tls = 0 listen = tls: 192.168.0.4:5061 tls_verify = 1 tls_require_certificate = 0 tls_method = SSLv23 tls_certificate = "/usr/local/etc/openser/user/user- cert.pem" tls_private_key = "/usr/local/etc/openser/user/user- privkey.pem" tls_ca_list = "/usr/local/etc/openser/user/user- calist.pem" --------------------------------------------------------------------------------------------------------------

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[Serusers] budgetone not registering
by ryan nalupa 07 Feb '06

07 Feb '06

hi all! can anyone help me with my little problem? i can't make my budgetone register to my ser box? i've used xlite softphones and they're able to call to/from each other. and they're quite fast loggin in. but when i used my budgetone, to test to my ser box, it won't register though i used the same configs on my softphones with the grandstream phone. is there any issue with ser and grandstream phones? ryan __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Deploying VoIP on a WAN
by Joao Pereira 06 Feb '06

06 Feb '06

Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we were using it anyway when we setup a (Asterisk+SER+SIP Proxy) Box to handle the "on-net dialout" calls. I'm now overwhelmed with the amount of SBCs that are suggested by the vendors to implement a solution. (http://www.juniper.net/solutions/literature/solutionbriefs/351085.pdf) Can anyone drop me some lines about this? I urgently need some feedback on this. Thanks! Joao Pereira www.fccn.pt

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[Serusers] Prepaid w/ SER
by Max Clark 06 Feb '06

06 Feb '06

Hi all, I have found a number of calling card applications for Asterisk which have been modified to support a prepaid sip service for origination/termination. Does something like this exist for SER? TIA, Max -- Max Clark http://www.clarksys.com

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[Serusers] SER and asterisk
by Sharon 06 Feb '06

06 Feb '06

hello all, I am able to route calls from asterisk to ser and vice versa. is it possible to make simultaneous calls from asterisk clients to ser clients without getting a 486 busy. -AA

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[Serusers] "admin" user y "heslo" password not valid in webser with presence
by victoria beltran 06 Feb '06

06 Feb '06

Hi, I have installed the presence snapshot and I am trying to install the last version of the serweb from the cvs (cvs -d:pserver:anonymous@cvs.serweb.berlios.de:/cvsroot/serweb checkout .). In the point 8, when I insert the "admin" username and "heslo" password on the page http://148.83.39.57/serweb/admin/admin.php, then "bad username or password" is written. The problem may be database structure has changed in presence snapshot. Did somebody achieve to use serweb or ctl tools on the presence snapshot? How can I create user credentials manually using mysql? I think this information is stored in "credentials" table, but I don't understand how I can insert data in it. thanks!! --------------------------------- LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

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[Serusers] TLS comments
by Klaus Darilion 06 Feb '06

06 Feb '06

Hi! I've tried the new TLS module: 1. It breaks compatibility with old TLS stack: Even when configured to use TLSv1, it sends an SSLv2 compatible HELLO: server2:~# ssldump New TCP connection #1: 10.10.0.41(33107) <-> 10.10.0.42(5063) 1 1 0.0088 (0.0088) C>S SSLv2 compatible client hello Version 3.1 I do not know if this is a problem with the new or the old stack. Further I do not know what other TLS enabled SIP products use. Do they accept SSL compatible HELLOs? 2. If there is an error during the TLS handshake (like above), ser keeps hanging without doing anything. IMO it shoud respond with error message (like it does when it can't establish a TCP connection): ser other proxy --INVITE--> <-100 ----- <-----TCP handshake----> --------TLS HELLO------> <---TCP RST ------------ ..... nothing happens ..... Instead I would expect: <-477 TLS error--- 00:21:41 server1 ser[3792]: ERROR: tls_server.c:275: IO error: (104) Connection reset by peer 00:21:41 server1 ser[3792]: ERROR: tcp_send: failed to send regards Klaus

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[Serusers] Asterisk Registering with SER question
by Ryan Pagquil 06 Feb '06

06 Feb '06

Hi, I've been registering asterisk to ser. I'm using SER as the outbound SIP trunk for Asterisk. Users registered with Asterisk will use the SIP trunk to reach SER registered users and PSTN's. Now when I register Asterisk with SER, on my SER's location table I see these record: Username Column = asterisk Contact Column = sip:s@202.84.24.47 I have a script running that checks the accounting records and sends BYE for the username that has no credit left. I found it hard doing this because of the record on the Contact column on location table of SER, everytime asterisk registers with SER. I could not send BYE to asterisk because of the broken contact information on Contact column of SER's location table. How can I correct this? Here is my sip.conf configuration: [general] port = 5060 bindaddr = x.x.x.x context = sip disallow=all allow=ulaw allow=alow fromuser=asterisk secret=test123 realm=mydomain.com register=>asterisk:test123@mydomain.com Thanks in advance, Ryan

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[Serusers] [serweb] Problem registering new user with SerWeb
by mkumar＠mantragroup.com 06 Feb '06

06 Feb '06

Hi All, I installed Serweb 0.9.4 and able to login as User and Admin. But when I hit on admin privileges it is not getting redirected to that page. Also I did not find any option to add a new user, is this a problem with my version of serweb or should I grant any new permissions to the admin. If I should give new permissions to admin then please send me what exactly the permissions are as I could not find them anywhere. Any help will be sincerely appreciated. Cheers, Manoj.

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[Serusers] "Round Robin outgoing calls"
by Jean-Michel Hiver 06 Feb '06

06 Feb '06

Hi, Is there a way with SER to forward calls which should go to PSTN to different gateways in a round robin fashion? Also, is there a way to dynamically disable / enable gateways without restarting ser? Your hints would be greatly appreciated. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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Re: [Serusers] firewall
by Toygun Mavinil 06 Feb '06

06 Feb '06

After my down question, as no replies come from list, my server hacked i though, Only local sipphones can register and no wan connection is allowed as my firewall allows all connections. Yesterday sipphones from wan could register but today none of them can. I see nothing on var/log/messages ???????????What do i have to do? Which ports i have to leave opento use ser with mediaproxy and sems? Toygun _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Toygun Mavinil Sent: Saturday, February 04, 2006 3:01 AM To: serusers(a)lists.iptel.org Subject: [Serusers] firewall Hi, I use ser 0.9.6 on fc4 Ser Works with no problem but Firewall is closed I use ser with mediaproxy and sems (trying) Which ports i have to left open while enabling the firewall? Toygun

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Re: [Users] forward_tcp
by Daniel-Constantin Mierla 06 Feb '06

06 Feb '06

In the OpenSER Core Cookbook should be, but the entry is not filled yet. http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook Cheers, Daniel On 02/03/06 20:37, Ramin Dousti wrote: > Thank you so much!!! > > Is that documented somewhere that I missed? > > Ramin > > On 2/2/06, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > >> Ramin Dousti wrote: >> >>> Hi Klaus, >>> >>> It seems that forward_tcp() requires at least one argument. Any other >>> suggestions? >>> >>> >> you can use: >> forward_tcp(uri:host); >> >> If you set the dst_uri via avpops, then it will have priority over >> request uri. >> >> Cheers, >> Daniel >> >> >>> BTW, when you said "create a new URI and URI contains a transport=tcp >>> parameter" what part of the URI contains the transport protocol? >>> >>> Ramin >>> >>> On 1/26/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: >>> >>> >>>> Hi Ramin! >>>> >>>> Please always cc to the list >>>> >>>> Ramin Dousti wrote: >>>> >>>> >>>>> Thanks, Klaus. >>>>> >>>>> Here is the full picture: >>>>> >>>>> I receive a SIP URI "sip:user@dest" on a UDP socket. The only thing I need to do >>>>> is to proxy this session on a TCP socket to "dest". >>>>> >>>>> I fugured I can do: >>>>> forward_tcp(dest); >>>>> >>>>> But when I extract dest into an AVP, forwasrd_tcp does not work with an AVP as >>>>> its argument. >>>>> >>>>> avp_printf("s:dst", "$td"); >>>>> forward_tcp("s:dst"); >>>>> >>>>> it tries to resolve "s:dst" literally. >>>>> >>>>> >>>> forward_tcp does not support pseudo variables. >>>> >>>> try forward_tcp() without parameters. >>>> >>>> regards >>>> klaus >>>> >>>> >>>> >>>>>> If the destination is stored in an AVP, use avp_pushto >>>>>> (http://openser.org/docs/modules/1.0.x/avpops.html#AEN291) to push the >>>>>> new URI into the request URI. >>>>>> >>>>>> Then use t_relay (make sure the URI contains a transport=tcp parameter). >>>>>> >>>>>> If the destination is static, use t_relay_to_tcp(ip, port) >>>>>> http://openser.org/docs/modules/1.0.x/tm.html#AEN320 >>>>>> >>>>>> regards >>>>>> klaus >>>>>> >>>>>> >>>>>> >>>>>> Ramin Dousti wrote: >>>>>> >>>>>> >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> I'm trying to forward a SIP message that's been received by UDP, to >>>>>>> $td through TCP. but the following configuration doesn't work: >>>>>>> >>>>>>> send_tcp("$td", 5060 ); >>>>>>> or >>>>>>> send_tcp($td, 5060 ); >>>>>>> or >>>>>>> avp_printf("s:dst", "$td"); >>>>>>> send_tcp("s:dst", 5060 ); >>>>>>> >>>>>>> Could you please help? >>>>>>> >>>>>>> -- >>>>>>> Ramin >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Users mailing list >>>>>>> Users(a)openser.org >>>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>> -- >>>>> Ramin >>>>> >>>>> >>>>> >>>>> >>> -- >>> Ramin >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> > > > -- > Ramin > >

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[Users] Re: OpenSER dispatcher module
by Daniel-Constantin Mierla 06 Feb '06

06 Feb '06

ds_select_dst() sets the outbound proxy address. There is an internal attribute that is used when sending the request. ds_select_domain() rewrites the domain part of request URI. Could you watch the network (ngrep -qt port 4050 on openser box) too see what is happening with the messages? Also, do you get any error in the syslog file (/var/log/syslog or /var/log/messages)? The way you use the dispatcher is ok. Cheers, Daniel On 02/03/06 00:48, Anders Brownworth wrote: > Daniel, > > I noticed a few posts on the OpenSER Users' list and thought I'd shoot > you a question directly as my subscription to the list doesn't seem to > be taking yet. > > I want to distribute SIP calls across a number of servers. (similar to > a load balancer) I intend to use the CallId hash algorithm to keep > subsequent messages going to the same server. > > I (like many others) am still trying to understand the dispatcher > module. Working off of the example configuration and running OpenSER > version 1.0.0, I have: > > ... > modparam( "dispatcher", "list_file", > "/usr/local/etc/openser/dispatcher.list" ) > #modparam( "dispatcher", "force_dst", 1 ) > > route { > if ( ! mf_process_maxfwd_header( "10" ) ) { > sl_send_reply( "483", "To Many Hops" ); > drop( ); > > }; > > ds_select_dst( "1", "0" ); > #ds_select_domain( "1", "0" ); > > forward( uri:host, uri:port ); > #t_relay( ); > > } > > > I have not been able to get it to do what I'm expecting. > > I have two different IPs in the dispatcher.list file and am expecting > the OpenSER instance to send an INVITE out to one of those IPs. With > debugging on a high level, I see the dispatcher module internally pick > one of those IPs but there is no attempt to send out an INVITE. > Throwing the t_relay( ) of course replies with a 100 trying, but still > no INVITE is sent out to one of the IPs in the destination set. > > Not really understanding the difference between ds_select_dst() and > ds_select_domain(), I tried ds_select_domain() with no change in > results. (both with and without t_relay()) I understand that > ds_select_destination() rewrites host and port, but then what does > ds_select_dst() do exactly? Why would you ever want to use > ds_select_dst()? It says ds_select_dst() "selects a destination from > the address set". Great, so then what does it do with that > destination? I am unclear on exactly what you are changing in the > request by running each of these functions. Can you clear that up for me? > > Then, once you get a destination, how do you tell OpenSER to send out > an INVITE to the destination server? Or does it not work this way? > > I also saw some talk about how ds_select_dst() "sets the destination > as outbound proxy" but how does that differ from the host / port > rewrite? I think I'm missing something here... > > My intent is to run a stateless distributer so I can massively scale > the front end of my setup. I don't think the tm module is an option > for what I want to do because scale is going to be key. > > I saw a mention of the lcr module but haven't taken a closer look at > that yet. I'd like to get this going as it seems like a very clean and > simple way to accomplish my goal. > > Any help you could provide would be much appreciated. > > Thanks, > > -Anders > > >

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[Users] minisip-tls registration with openser problem - new devlopment
by Pjothi 06 Feb '06

06 Feb '06

Dear all, A new development in my attempt to register minisip with openser in tls mode. Previously I was getting an exception and after SSL_CTX but now the socket is getting created but still the client is not registered. I am fairly confident about the Certificates, since I created both client and sever certificate with the script files in OpenSER source. note: minisip (latest source from svn) and OpenSER (compiled with TLS support) both runs in a LAN environment, both on Suse Linux 10 Below, is a post of minisip start. ---------------------------------------------------------------------------------------------------------------------- user1@linux:~> minisip Starting MiniSIP ... welcome! Initializing NetUtil Creating SipSoftPhoneConfiguration init 1/9: Creating timeout provider init 2/9: Creating GUI Creating GTK GUI (minisip:6125): gtkmm-WARNING **: gtkmm: Attempt to call Gtk::manage() on a Gtk::Window, but a Gtk::Window has no parent container to manage its lifetime. Minisip: gtk 1 Minisip: gtk 2 Setting contact db Thread 2 running - doing initParseConfig init 3/9: Parsing configuration file () Config file version checked ok! SipIdentity::SipIdentity : cretated identity id=1 SipIdentity::setSipUri: sipUsername=<user4> sipDomain=<192.168.0.4> SipIdentity::setSipProxy: autodetect is false; userUri=user4(a)192.168.0.4; transport = TLS; proxyAddr=192.168.0.4; proxyPort=5061 SipProxy:setProxy(str) : addr = 192.168.0.4 SipIdentity::setProxy: manual sipproxy success ... SipIdentity::setProxy: else ... Identities: identity=1; username=user4; domain=192.168.0.4 proxy=[proxyString=192.168.0.4; proxyString=192.168.0.4; port=5061; transport=TLS; autodetect=no; user=user4; password=user4; expires=1000]; isRegistered=0 init 4/9: Creating IP provider SimpleIPProvider: localIp = SimpleIPProvider: checking interface = lo with IP=127.0.0.1 SimpleIPProvider: checking interface = eth0 with IP=192.168.0.3 SimpleIPProvider: using localIP = 192.168.0.3 init 5/9: Creating MediaHandler Sound I/O: using Spatial Audio Mixer Adding audio codec: G.711 init 6/9: Creating MSip SIP stack init 7/9: Connecting GUI to SIP logic init 8.2/9: Starting TCP transport worker thread init 8.3/9: Starting TLS transport worker thread init 9/9: Registering Identities to registrar server Registering user user4(a)192.168.0.4 to proxy 192.168.0.4, requesting domain 192.168.0.4 SipMessageTransport: sendMessage: creating new socket Creating new SSL_CTX SipMessageTransport: sendMessage: reusing old socket TLS: Shutting down TLS Socket ------------------------------------------------------------------------------------------------------------------ The above is the message I get when I start minisip and I cannot understand the reason why its not registering with the OpenSER. When I disable TLS in OpenSer and connect minisip in udp mode, it registers perfectly with openser. would be very helpful if you can give me your thoughts.....thank you very much for your time. regards, Pjothi

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[Serusers] postgresql.conf with Radius
by Voipers Portugal 06 Feb '06

06 Feb '06

Hi, Does anyone has an accounting mechanism working with SER and Radius? I have it all working ok, but when the start and stop request comes to Radius server, it does not write it to the postgres BD. Could anyone provide or let me know where i can find the postgresql.conf file that is supposed to be in the Radius Server machine? Regards, Jose Simoes

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[Users] SIP Session Timeout
by martin＠campus-merseburg.de 06 Feb '06

06 Feb '06

Hi, could i add a Session-Expires header in an invite-request so that the sip session has the sip session timer support? ciao, martin

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Re: Re: [Serusers] radius with ser
by Kenny Yeh 06 Feb '06

06 Feb '06

Greger V. Teigre,Hi I got a new debug information below: Feb 6 16:48:14 debian3 ser: ERROR: auth_radius: can't get code for the Sip-URI-User attribute Feb 6 16:48:14 debian3 ser: init_mod(): Error while initializing module auth_radius what do you think about those? Kenny ======= 2006-02-06 17:53:27 您在来信中写道：======= >See further above in your log. You probably have an error when ser tries to >load the radius module. Probably because the radiusclient library is not >found. >g-) >----- Original Message ----- >From: "Kenny Yeh" <kenny(a)artdio.com.tw> >To: <serusers(a)iptel.org> >Sent: Monday, February 06, 2006 10:42 AM >Subject: [Serusers] radius with ser > > >> Hi, >> >> I have a big problems.I followed radius howto file where I found on >> iptel.org,and tested freeradius.when I run ser start,I found some error in >> syslog below: >> >> Feb 6 03:36:39 debian ser: find_export: <radius_www_authorize> not >> found >> Feb 6 03:36:39 debian ser: find_export: <radius_www_authorize> not found >> Feb 6 03:36:39 debian ser: parse error (137,36-37): unknown command, >> missing loadmodule? >> >> I loaded auth_radius.so from ser.cfg,why ser cann't found >> radius_www????? >> >> who can help me!!!! thx!!! >> >> >> Kenny >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)iptel.org >> http://mail.iptel.org/mailman/listinfo/serusers >> > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-02-06

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[Users] TOS setting for SIP signalling
by Matteo Piazza 06 Feb '06

06 Feb '06

Hi, I don't understand if the Tos setting is insert in the CVS of 1.0.0 version or it is in the devel branch. Thank's Matteo

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Re: [Users] Problems registering NAT clients
by Jesus Rodriguez 06 Feb '06

06 Feb '06

On Mon, 6 Feb 2006, Paco Orozco wrote: Hola Paco, I Cc: to the list. >> You can see here that the NAT device is changing the source port from >> 5060 to 10145, which is the port used by Openser to send the reply. >> Openser is doing fine while your NAT device is doing wrong because it >> keeps the external rewriten source port when sending the packet to the >> internal side of the NAT (see the second line of the above tcpdump >> trace). > > You are right, I was confused because I thought that Openser was > answer me wrong. > >> Which NAT device/router are you using?. Are you forwarding ports from >> the external to the internal side at the NAT device?. > > My NAT device is a 3Com Office Connect 812. A typical ADSL router. I > haven't got any forwarding ports external to internal? > > How can i solve this? Upgrade the firmware of the router. That model is already known because the NAT management problems for UDP packets. With the last firmware version the problem should be fixed. Saludos JesusR. ------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------

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[Serusers] radius with ser
by Kenny Yeh 06 Feb '06

06 Feb '06

Hi, I have a big problems.I followed radius howto file where I found on iptel.org,and tested freeradius.when I run ser start,I found some error in syslog below: Feb 6 03:36:39 debian ser: find_export: <radius_www_authorize> not found Feb 6 03:36:39 debian ser: find_export: <radius_www_authorize> not found Feb 6 03:36:39 debian ser: parse error (137,36-37): unknown command, missing loadmodule? I loaded auth_radius.so from ser.cfg,why ser cann't found radius_www????? who can help me!!!! thx!!! Kenny

2 1

[Serusers]Problem with AS5300 to SER
by 昂達 06 Feb '06

06 Feb '06

Hello : This is my AS5300 DID(direct-inward-dial) configuration *********************************************** ! dial-peer voice 101 pots incoming called-number 41000.T no digit-strip direct-inward-dial port 1:D forward-digits all ! dial-peer voice 102 voip huntstop application session destination-pattern 41000.T rtp payload-type nte 98 voice-class codec 99 session protocol sipv2 session target sip-server dtmf-relay rtp-nte ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers expires 300000 timers connect 300 sip-server ipv4:10.0.0.1 ! **************************************************** I wanna call AS5300 to SER , I use "csim start [DID number]" command but it show this error message: csim err csimDisconnected recvd DISC cid(503) csim: loop = 1, failed = 1 csim: call attempted = 1, setup failed = 1, tone failed = 0 and I use " show call history voice " command I find that : DisconnectText=bearer capability not authorized (57) Do you see any other solution for this problem? Regards, Hunter

1 0

[Users] record_route_preset(): No memory left
by Bastian Schern 05 Feb '06

05 Feb '06

Hello to all, I'm using the latest CVS 1_0_0 and I got the following message in the log after a "record_route_preset( "213.191.xxx.xxx:5060;nat=yes" )": record_route_preset(): No memory left Why this comes? Regards Bastian ____________ Virus checked by G DATA AntiVirusKit Version: AVK 16.5138 from 28.01.2006 Virus news: www.antiviruslab.com

2 17

[Serusers] rtpproxy and fax clients under linux nat
by Antanas Masevicius 04 Feb '06

04 Feb '06

Hello, latetly i got a bit confused while playing with rtpproxy. It is said that it supports nat clients with symmetric nat. As i know, linux is also symmetric by default. Now i get into situation where my natted clients work for voice, but does not for fax. I traced that my nat does not forward incomming T38 stream saying "destination unreachable (port unreachable)". After looking at ip_conntrack during audio call i see: udp 17 180 src=192.168.0.46 dst=my.rtpproxy.ip sport=4004 dport=19006 src=my.rtpproxy.ip dst=my.external.ip sport=19006 dport=1024 [ASSURED] use=1 udp 17 29 src=192.168.0.46 dst=my.rtpproxy.ip sport=4005 dport=19007 [UNREPLIED] src=my.rtpproxy.ip dst=my.external.ip sport=19007 dport=4005 use=1 udp 17 171 src=192.168.0.46 dst=my.rtpproxy.ip sport=5069 dport=5060 src=my.rtpproxy.ip dst=my.external.ip sport=5060 dport=5069 [ASSURED] use=1 but looking at first line it doesn't seem like symmetric nat, it is clear that packets out of my.external.ip goes with low (1024) sports. Should it be like so or my linux masquarading is misconfigured? This audio session work ok for some reasons unknown to me. Can anyone explain the behaviour? >From second line we can see that packect can't reach 4005 port. Similar problem prevents me from receiving t38 stream i guess. How do you configure your linux natting for set up with rtpproxy? Antanas NTT

1 0

[Users] Problems registering NAT clients
by Paco Orozco 04 Feb '06

04 Feb '06

Hello, First of all, I'm spanish and my english isn't good, so please excuse my mistakes. I've been working with openser for a year. Now I'm modifying my SIP service in order to allow clients in NAT scenarios. I've read about symmetric / full cone / ... and other NAT scenarios. I've read about mediaproxy / rtpproxy and nathelper modules. I've got a openser.cfg working properly without NAT clients. And I've modified it (see attach). Client X-Lite: IP: 192.168.1.2 (public address 81.39.220.5) Listen SIP port: 5060 Listen RTP port: 8000 Server: Openser 1.0 with Media Proxy IP: 147.83.197.39 Config in opernser.cfg If i try to register with a UAC (X-Lite or K-Phone) with symmetric NAT scenario. Opernser server doesn't answer the client to 5060 port, as i need. X-Lite Diagnostic LOG is client_messages.txt Client timed out without receiving any server messages. This is a tcpdump on clients side: 17:42:23.458980 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:23.558019 IP 147.83.197.39.5060 > 192.168.1.2.10145: SIP, length: 538 17:42:23.558077 IP 192.168.1.2 > 147.83.197.39: ICMP 192.168.1.2 udp port 10145 unreachable, length 556 17:42:25.085485 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:28.108960 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:32.643903 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 2 17:42:34.156151 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 On the server this is the tcpdump: 17:42:23.904035 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 17:42:23.904261 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 538 17:42:23.905028 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 680 17:42:24.004623 IP 81.39.220.5 > 147.83.197.39: ICMP 81.39.220.5 udp port 10145 unreachable, length 556 17:42:25.525470 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 17:42:25.525607 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 538 17:42:25.526314 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 680 17:42:28.549322 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 What's wrong? It's necessary to use a STUN server? Why can't register? Why openser doesn't reply to 5060 port? My client it's only reachable in this. Can anyone help me? Anyone have a OpenSer config working? Thanks in advance -- ------------------------------------- Aquel que planta árboles bajo los que sabe muy bien que nunca se sentara, ha descubierto el auténtico significado de la vida.

2 1

[Users] Problems registering NAT clients
by Paco Orozco 04 Feb '06

04 Feb '06

Hello, The last message was mailformed.... sorry First of all, I'm spanish and my english isn't good, so please excuse my mistakes. I've been working with openser for a year. Now I'm modifying my SIP service in order to allow clients in NAT scenarios. I've read about symmetric / full cone / ... and other NAT scenarios. I've read about mediaproxy / rtpproxy and nathelper modules. I've got a openser.cfg working properly without NAT clients. And I've modified it (see attach). Client X-Lite: IP: 192.168.1.2 (public address 81.39.220.5) Listen SIP port: 5060 Listen RTP port: 8000 Server: Openser 1.0 with Media Proxy IP: 147.83.197.39 Config in opernser.cfg If i try to register with a UAC (X-Lite or K-Phone) with symmetric NAT scenario. Opernser server doesn't answer the client to 5060 port, as i need. X-Lite Diagnostic LOG is client_messages.txt Client timed out without receiving any server messages. This is a tcpdump on clients side: 17:42:23.458980 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:23.558019 IP 147.83.197.39.5060 > 192.168.1.2.10145: SIP, length: 538 17:42:23.558077 IP 192.168.1.2 > 147.83.197.39: ICMP 192.168.1.2 udp port 10145 unreachable, length 556 17:42:25.085485 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:28.108960 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:32.643903 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 2 17:42:34.156151 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 On the server this is the tcpdump: 17:42:23.904035 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 17:42:23.904261 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 538 17:42:23.905028 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 680 17:42:24.004623 IP 81.39.220.5 > 147.83.197.39: ICMP 81.39.220.5 udp port 10145 unreachable, length 556 17:42:25.525470 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 17:42:25.525607 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 538 17:42:25.526314 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 680 17:42:28.549322 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 What's wrong? It's necessary to use a STUN server? Why can't register? Why openser doesn't reply to 5060 port? My client it's only reachable in this. Can anyone help me? Anyone have a OpenSer config working? Thanks in advance -- ------------------------------------- Aquel que planta árboles bajo los que sabe muy bien que nunca se sentara, ha descubierto el auténtico significado de la vida.

1 0

[Users] user is not found
by Brijesh 04 Feb '06

04 Feb '06

Hi all I am new to OpenSER. I am using one of sip client. I want to connect sip clients through openSER proxy I have installed it and configured the clients but giving me error that user is not found Can anyone tell me what is reason for it ?

1 0

[Serusers] Problem with ACK and BYE..
by pankaj.munjal＠wipro.com 04 Feb '06

04 Feb '06

Hi everyone, I've configured ser to work with semstalkflite module of sems. This is how it works: A user calls up sip:READOUTTHISTEXT@ser and ser should be able to play back "READ OUT THIS TEXT" using a media server(sems). Now it seems to be working fine, but, after having a INVITE-200OK-ACK dialog b/w ser and caller "A", ser is not absorbing the ACK. i,e, for some reason, Caller "A" keeps sending ACK again and again. FYI.. caller "A" uses X-lite. Here is the part of the ser config file that i've modified: if (method=="ACK"){ log(1,"Pankaj --> I've hit ACK!! "); # absorb ACKs break; }; Isn't it good enuf to absorb the incoming ACKs? Anyone with any pointers?? Thanks in advance, Pankaj Munjal The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[Serusers] firewall
by Toygun Mavinil 04 Feb '06

04 Feb '06

Hi, I use ser 0.9.6 on fc4 Ser Works with no problem but Firewall is closed I use ser with mediaproxy and sems (trying) Which ports i have to left open while enabling the firewall? Toygun

1 0

Re: [Users] Using uac_replace_from to munge username between ser and Asterisk?
by Bogdan-Andrei Iancu 03 Feb '06

03 Feb '06

if you enable auto from_restore_mode, you do not need to perform any restore from script. Just replace the from in the initial INVITE and this is it - all replies and sequential request would be auto fixed (restore/replace). regards, bogdan Barry Flanagan wrote: > Bogdan-Andrei Iancu wrote: > >> Hi Barry, >> >> have you set auto from restoring? See: >> http://openser.org/docs/modules/1.1.x/uac.html#AEN75 > > > Yes, but I am not sure where it is supposed to go. > > I have the following in just before relaying to Asterisk: > > rewritehostport("XXX.XXX.XXX.XXX:5060"); > uac_replace_from("$fn","sip:$au_$ar@$fd"); > append_hf("P-hint: GATEWAY\r\n"); > t_relay("udp:XXX.XXX.XXX.XXX:5060"); > > > and I put in uac_restore_from(); just after the record_route() > > > with all the other modparams I have: > > modparam("uac","from_restore_mode","auto") > > > Thanks for the help. > > -Barry > > >> regards, >> bogdan >> >> Barry Flanagan wrote: >> >>> >>> So, the only way around it that I can see is to somehow have OpenSER >>> change the username to username_domain so that each will be unique. >>> >>> It looks like uac_from_replace should handle this. I have tried it, >>> and I can see that Asterisk does in fact get user_domain@domain in >>> the first invite, but thereafter for some reason OpenSER changes it >>> to just _@domain for subsequent requests. >>> >>> >>> Regards, >>> >>> -Barry >>> > >

2 5

[Users] Receiving different VIA in INVITE and CANCEL
by Papadopoulos Georgios 03 Feb '06

03 Feb '06

Hello, I have an issue with some user agents, (Linksys PAP2 and Sipura 3000) sending different Via in CANCEL than in INVITE. Of course the result is that the caller hangs up and the callee keeps ringing. I tried to use modparam("tm", "via1_matching", 0) but it did not solve the problem. I have two questions: 1. Is the via1_matching in TM working as described or is it broken? 2. What can be causing this behavior of the user agent? Not all our users who have these user agents report this problem (actually very few have the problem). In our lab we tried same user agent, same firmware but could never reproduce this behavior no matter how bad we tried to mess up their configuration. Any help will be greatly appreciated. Thanks. George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Serusers] No comunication between user through SER. S.O.S.
by Ernesto Garcia 03 Feb '06

03 Feb '06

Hi!!, I'm very new to this topic, I'm building a call center and I testing with Asterisk and SER, the problem is that when some registered user call to other user tha is registered too the phone of it don't detect the incoming call. Plase, what well be is the problem ? Ernesto

1 0

Re: [Serusers] high-availibility setup using f5 bigip
by Jack Wei 03 Feb '06

03 Feb '06

I want to set up 2 SER boxes and 2 Asterisk boxes using 192.168.1.X IPs. I set up 2 VIPs, 1 for SER and 1 for Asterisk. SER redirect voice mail calls to Asterisk VIP. However, if you look at the SIP logs, Asterisk is using a 192.168.1.X IP for the contact address. Also, there is no audio since the SDP is using the 192.168.1.X IP. I've set the externip and localnet in Asterisk but nothing happened. My goal is to set up a load-balancing pools of SER and Asterisk behind a BIGIP switch. --- Charles Wang <lazy.charles(a)gmail.com> wrote: > I think that the range of this question is too large. > You should tell us what your scenario is. And tell us more about your > configurations. > > 2006/2/2, Jack Wei <cowlemon(a)yahoo.com>: > > hi, > > > > I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do > > load-balancing. I'm using Asterisk as a voicemail application only and > have > > successfully integrated SER with Asterisk without the switch. But when I > try > > to use the switch as a load-balancer, I get lots of NAT problems. Does > anyone > > know how to setup the switch and SER/Asterisk properly? > > > > Thanks, > > Jack > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- > > Best Regards > Charles > Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] serctl, mysql and presence snapshot
by victoria beltran 03 Feb '06

03 Feb '06

>The reason is, that in this version (or some little older) has changed >the database structure and serctl is broken there (most serctl commands >are unusable). > >Vaclav Then, I believe the serweb will not work successful either with the presence snapshot, Is this right? Thus, neither the serweb nor the serctl tool works. Therefore, Are mysql commands the single way to manage (i.e create and drop users) the ser server?? thanks! --------------------------------- LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

2 1

[Serusers] serctl, mysql and presence snapshot
by victoria beltran 03 Feb '06

03 Feb '06

2 1

[Serusers] Problems with DTMF using SER and Asterisk
by Anders Hansson 03 Feb '06

03 Feb '06

Hi all, I have a problem getting dtmf tones throgh to PSTN when using asterisk. The connections is SIP phone - Asterisk - SER - Cisco GW - PSTN When dialling PSTN numbers from the SIP phone and pressing the * or # button nothing is happening. I' ve set the dtmfmode=rfc2833 in asterisk. The problem is that to be able to control the asterisk voicemail the dtmfmode has to be =rfc2833, and the I can control the voicemail. But when calling external to PSTN the * and # button is not working. I do also have another problem, and that is when connecting a ATA box to the Asterisk. If I set dtmfmode=inband, voicemail in asterisk can be controlled but not the calls to external PSTN numbers. If I set dtmfmode=rfc2833, voicemail in asterisk do not work, but external calls to PSTN can be controlled with * and #. What can be the error? Please help, Thanks Anders

1 0

[Serusers] extcmd
by Jose Silva 03 Feb '06

03 Feb '06

Hi, is there any documentation on the extcmd module? Thanks in advance Joe

3 2

RE: [Serusers] Correct SER version to download for RedHat Enterprise
by Bart J. Smit 03 Feb '06

03 Feb '06

There are some RPM's for 0.9.6 here: http://ftp.upjs.sk/pub/users/sal/Fedora/4/voip/ Note that ser-mysql requires MySQL >= 4.1. Your distro may have MySQL 3.x. Bart... -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Paul Hazlett Sent: 03 February 2006 12:31 To: Henry Meyer Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Correct SER version to download for RedHat Enterprise Henry, Your best bet, IMHO, is to check out the rel_0_9_0 branch from the Berlios CVS repo. Even though the branch name indicates version ser-0.9.0, it is really ser-0.9.x, which I believe the latest is 0.9.6. This is the most recent stable version of SER. Regards, Paul On 2/2/06, Henry Meyer <Henry.H.Meyer(a)hofstra.edu> wrote: > I'm looking to install SER on a server running Red Hat Enterprise Linux > AS-up (2.4.21-37.EL). What do I need and where do I download it? > Thanks > > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Correct SER version to download for RedHat Enterprise
by Henry Meyer 03 Feb '06

03 Feb '06

I'm looking to install SER on a server running Red Hat Enterprise Linux AS-up (2.4.21-37.EL). What do I need and where do I download it? Thanks

2 1

[Serusers] Problem integratin Asterisk 1.2.4 and SER 0.9.6
by ernesto＠serverlinux.aroasystem.cl 03 Feb '06

03 Feb '06

Hi all; I'm trying to integrate asteris and OpenSER and I don't get any result, I get communication well with asterisk but I install OpenSER for avoid the problem with NAT. In my extension.conf I write this SERADDRESS=1.2.3.4:5062 exten => 88884,1,Dial(SIP/88884@${SERADDRESS},20,r) exten => 88885,1,Dial(SIP/88885@${SERADDRESS},20,r) in sip.conf [general] context=openser bindaddr=192.168.0.10 bindport=5060 realm=mydomain.com .... [Openser] type=peer context=openser host=1.2.3.4 port=5062 insecure=very in openser.cfg I use one of the examples: # # $Id: flag_reply.cfg,v 1.2 2005/07/25 16:27:33 miconda Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 fifo="/tmp/ser_fifo" listen=1.2.3.4 port=5062 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ setflag(1); t_relay_to_udp("1.2.3.4", "5060"); return; } my softphone is eyeBean. Now the situation is this. my softphones get registered, but when one phone call the other this don't receive the call. in the database od openser the two client are registered. Please I need a help, excuse me with my english, is't not my mother tongue. Thanks in advance, Ernesto.

1 0

[Serusers] Serweb
by Ismael 03 Feb '06

03 Feb '06

Hi, I wrote on my browser http://192.168.1.68/serweb/admin/index.php but it produces an 404 error. Whereas my y apache server, have on this configuration file an alias : /serweb /var/lib/serweb/html I don't understand ! :-[ Someone could help me ? Thanks, Ismaël

1 0

Re: [Serusers] Monitoring
by Cesc 03 Feb '06

03 Feb '06

Hi, some asked me if i could share the script ... yes, i can ... but it is in no way a nice script. It is done by me (mostly) ... but is done FOR me, so you may need to change a few things to make it work ... sorry :) Cesc PS - there is more info on the README inside the attached tgz file On 1/27/06, Hakan YASTI <hakanyasti(a)gmail.com> wrote: > Hi Cesc, > > Could you send the your script that checks the ser activity. What I do is to > check the ser processes via ps -ax but as I see your solution is better. > Thanks, > > ----- Original Message ----- > From: "Cesc" <cesc.santa(a)gmail.com> > To: <hgaillac-sip(a)yahoo.fr> > Cc: <serusers(a)lists.iptel.org>; "Atle Samuelsen" <clona(a)cyberhouse.no> > Sent: Friday, January 27, 2006 3:40 PM > Subject: Re: [Serusers] Monitoring > > > As answered, no. > > What i did to monitor ser and sems was to create bash scripts which > restart ser and/or sems in case of failure. > > For ser, the script sends fake REGISTER messages (with expires=0). If > the answer is 200 OK (or a 40x proxy authenticate) ... then ser is > alive. If nothing is read, then i restart ser. > For sems, i tried two approaches. > - active: send a specially crafted message (INVITE) so that you know > for sure that ser will route it to SEMS ... depending on the answer, > you should be able to know wether SEMS is alive or not ... but this > has a problem (see below for a larger explanation). > - passively: start sems so it writes its PID to a log file ... then > monitor (using ps aux) that pid ...if not there, restart SEMS. > - i seem to remember someone mentioned a status module for sems, which > would allow you to send commands via the fifo/unixsock interface, and > check actively sems (including activity statistics) ... is this true? > > ===== ACTIVE MONITORING PROBLEM > What i tried to do is to created unique INVITE messages, which i make > sure that ser routes to sems. They had unique call-ids and to/from > tags ... thus theoretically ser should see them as independendent > requests. > But, if sent these INV requests too often (that is, less than a minute > or so ... time enough for the transaction to be destroyed in ser mem, > i guess ... ), ser would mistakenly think they were retransmissions > ... > It would also start acting weird and sending replies to the wrong port ... > This is why i went to passive monitoring ... i do not have time to > investigate any further ... but i can provide the script for someone > else to try ... > BTW, i use ser 0.9.6 > > Regards, > > Cesc > > > > On 1/27/06, hgaillac-sip(a)yahoo.fr <hgaillac-sip(a)yahoo.fr> wrote: > > Yes I know about snmp module. > > There is no way to monitor SER vi snmp !? > > > > Harry > > --- Atle Samuelsen <clona(a)cyberhouse.no> a écrit : > > > > > Hi Harry, > > > > > > ser does not support SNMP. There is a module, but it > > > has not been > > > maintained for ages and does not work. > > > > > > -Atle > > > > > > * hgaillac-sip(a)yahoo.fr <hgaillac-sip(a)yahoo.fr> > > > [060127 13:05]: > > > > Hi asterisk and ser users, > > > > > > > > Is there a solution to monitor asterisk and ser > > > with > > > > snmp ? > > > > > > > > Regards > > > > Harry > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > > > > > Nouveau : t?l?phonez moins cher avec Yahoo! > > > Messenger ! D?couvez les tarifs exceptionnels pour > > > appeler la France et l'international. > > > > T?l?chargez sur http://fr.messenger.yahoo.com > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les > > tarifs exceptionnels pour appeler la France et l'international. > > Téléchargez sur http://fr.messenger.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Serusers] Problem with BYE Signal
by Asep Saifudin 03 Feb '06

03 Feb '06

Hi All, I have problem with BYE Signal. Here my scheme : My SIPPhone device--->SER--->SIP to H323 Converter--->Softswitch--->Quintum FXO Gateway===>Phone line===>Telco Provider===>My Destination Number. (Note: ---> meaning is via IP & ===> is wireline). I have make several testing call (e.q to the same destination number) but the problem is sometime if I finish my call, the CDR logged to ACC table and sometime didn't Logged. Why ? Between Caller and Calle which one should be responsible to give BYE Signal ? or should be both of them ? Thanks & Regards, ~Asep -Jakarta

2 1

[Serusers] 503 Service Unavailable processing
by Ramin Nikaeen 02 Feb '06

02 Feb '06

Friends, Can anyone point me in the right direction with respect to processing 503 (service unavailable) replies?! I did some testing and here is what I have found so far: 1. I expected the 503 to be received in failure_route but, contrary to my expection, I am catching it in onreply_route 2. I tried to relay the call within onreply_route to an IVR but the t_relay fails giving the message: DEBUG: w_t_relay entered! 12(20326) ERROR: w_t_relay_to: unsupported mode: 2 ERROR: w_t_relay_to: unsupported mode: 2 3. I tried processing the 503 upon receipt of next message at the top of routing configuration but didn't work: a) I set an avpop in onreply_route upon receipt of 503 and checking it at the top of my configuration in the next round but the avp_check returns false b) I set a flag in onreply_route upon receipt of 503 and checking it at the top of my configuration in the next round but isflagset returns false c) I set a gflag in onreply_route upon receipt of 503 and checking it at the top of my configuration in the next round but isgflag_set returns false Any suggestions would be very much appreciated. Thanks a bunch ramin

1 0

[Serusers] Checking of Destination IP
by Atle Samuelsen 02 Feb '06

02 Feb '06

Hi Guys, is there a way to check the Destination address of a Request, before it get's t_relay(); ? I think it's possible with branch-routes, but that's only in cvs head :( -Atle

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[Serusers] Serweb installation
by Ismael 02 Feb '06

02 Feb '06

Hello, where i specify the hostname server ? On the 8th step of the serweb installation, it is wrote : "Open the file http://<your-host><your-install-dir>/admin/index.php" What do you mean by <your-install-dir> ? For the moment, i tried with : http://192.168.1.71/var/libserweb/html/admin/index.php, but it doesn't function ?? Could you help me ? Thanks, Ismaël

2 1

[Users] DNS CNAME query in openser?
by Greg Fausak 02 Feb '06

02 Feb '06

Can I do a cname or txt query in openser? I'm trying to create a cname for a enum soa. That is, I want to do: soa: e164.test.com cname: other.test.com -> e164.test.com Is there any way to accomplish this sort of thing through DNS? -g -- Greg Fausak greg(a)thursday.com

2 3

[Serusers] SER & SEMS on separate machines
by Amos M. Nungu 02 Feb '06

02 Feb '06

Hello members, I have SER and SEMS on the same machine and it works fine. For performance reasons, I would like to them to be on separate machines. My query is the fact that SEMS uses the email address from the subscriber table in SER db. Do I need to have SER db on the machine where SEMS is running (means 2 identical dbs) or how can I achieve this? Regards, Amos.

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[Users] INVITE transaction timer vs. SIP Proceeding timer
by Arek Bekiersz 02 Feb '06

02 Feb '06

Dear all, I think it will be useful for everybody. I have discussion with SIP device manufacturer about expiration time of call attempt. When call is made from UA it sends INVITE with "Expires: 65". It is controlled by device's timer setting "ring_time_limit". This timer can be set to anything between 10 and 600 seconds, we set it to 65 seconds. Fine. However at the same time it starts a special timer, say "max_response_time_invite". It can be set to anything between 8 and 20 seconds. We set it to 20 sec. Fine so far. CASE 1: Imagine we call from UA to PSTN number, using one of VoIP->PSTN providers. We have more than one provider, so we setup SER with "fr_timer" value of, let's say 25 seconds and "fr_inv_timer" of 60. We prepare failure routes in case when those timers hits in PSTN call. When "fr_timer" hits, we simply reroute to another gateway. Great. SER forwards INVITE from UA to remote PSTN gateway and at the same time sends back "100 Trying" to UA. Fine. Imagine now that it is busy-hour and it takes PSTN provider 21 seconds to send back "180 Ringing" (or "183 Session Progress") and after 8 seconds more remote callee picks up ("200 OK"). Unfortunately device wil send CANCEL to SER, because it hasn't received "180 Ringing" or "183 Session Progress" within the "max_response_time_invite" setting of 20 seconds. It only received "100 Trying" at the beginning. This ruined any failure route attempts from SER, as call failed. CASE 2: Imagine now a second call. This time PSTN provider send "183 Session Progress" after 14 seconds and remote callee picked up after further 15 seconds. This time everything went good. UA did not hit "max_response_time_invite" timer - it waited patiently for whole 29 seconds for remote callee to pick up. I think that when UA receives provisional response (like "100 Trying") from SER it should stop timer "max_response_time_invite" and use 'ring_time_limit'. What device does now, it uses: * "max_response_time_invite" when "100 Trying" received * "ring_time_limit" when "180 Ringing" or "183 Session Progress" received I haven't really found explanation of this in RFC3261. When you look at SIP state machine on page 128, it is not explicitly said which timer controls behaviour of UA when in "Proceeding" state ("Proceeding" is entered oin receipt of ANY provisional response). QUESTION 1: what timer controls behaviour of UA when in "Proceeding" state? Especially regarding timeout. Is this still the 'timer B' started at the beginning of INVITE transaction? If it is 'timer B', why UA is CANCELing transaction after 20 seconds, if it informed by "Expires: 65" field in INVITE that it set 'timer B' to 65 seconds. QUESTION 2: What should I propose to manufacturer for maximum value for "max_response_time_invite"? -- Thanks, Arek Bekiersz

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Re: [Users] forward_tcp
by Klaus Darilion 02 Feb '06

02 Feb '06

Hi Ramin! Please always cc to the list Ramin Dousti wrote: > Thanks, Klaus. > > Here is the full picture: > > I receive a SIP URI "sip:user@dest" on a UDP socket. The only thing I need to do > is to proxy this session on a TCP socket to "dest". > > I fugured I can do: > forward_tcp(dest); > > But when I extract dest into an AVP, forwasrd_tcp does not work with an AVP as > its argument. > > avp_printf("s:dst", "$td"); > forward_tcp("s:dst"); > > it tries to resolve "s:dst" literally. forward_tcp does not support pseudo variables. try forward_tcp() without parameters. regards klaus >>If the destination is stored in an AVP, use avp_pushto >>(http://openser.org/docs/modules/1.0.x/avpops.html#AEN291) to push the >>new URI into the request URI. >> >>Then use t_relay (make sure the URI contains a transport=tcp parameter). >> >>If the destination is static, use t_relay_to_tcp(ip, port) >>http://openser.org/docs/modules/1.0.x/tm.html#AEN320 >> >>regards >>klaus >> >> >> >>Ramin Dousti wrote: >> >>>Hello, >>> >>>I'm trying to forward a SIP message that's been received by UDP, to >>>$td through TCP. but the following configuration doesn't work: >>> >>> send_tcp("$td", 5060 ); >>>or >>> send_tcp($td, 5060 ); >>>or >>> avp_printf("s:dst", "$td"); >>> send_tcp("s:dst", 5060 ); >>> >>>Could you please help? >>> >>>-- >>>Ramin >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > > > -- > Ramin > >

3 3

[Users] How to check Displayname in From Tag
by martin＠campus-merseburg.de 02 Feb '06

02 Feb '06

Hi, i want to change the "From:" tag with uac_replace_from when the displayname is "anonymous". with from_uri i can only check the uri. is there a way to check the displayname in the "From: " tag? for example: From: "Anonymous <sip:1234@blah.net>" i want to change it to: From: "Anonymous <sip:Anonymous@Anonymous>" Ciao

4 7

[Serusers] Serweb
by Ismael 02 Feb '06

02 Feb '06

hello, i have followed serveb installation instructions, and know i am on the last step : "Open the file http://<your-host>/<your-install-dir>/admin/index.php" But i don't understand what i have to put on "your-host" ? Thanks, Ismaël

2 1

[Serusers] Install and configure SER
by Ismael 02 Feb '06

02 Feb '06

Hello, I want to install SER on my debian. Could you explain me step by step how to install and configure ser server on my computer ?? How attach mysql to SER server ? Thanks to advance, Ismaël

1 0

[Serusers] Need help to install SER and Serweb
by Ismael 02 Feb '06

02 Feb '06

Hello, I installed Debian on my computer, and now i want to install ser proxy ans serweb. To install SER, i did : // installation of apache, mysql, and phpmyadmin apt-get install apache mysql-server phpmyadmin // installation of SER dpkg -i ser_0.9.6-0.1_i386.deb // restart of SER service /etc/init.d/ser restart To install serweb, i just did : tar -xvzf serweb-0.9.4.tar.gz What do i have to do now ?? Thanks for your help, Ismaël

2 1

[Serusers] TCP sockets and NATs
by Teemu Harju 02 Feb '06

02 Feb '06

Hi, This is a question that has been bugging me for some time now. If I use SER and the clients connected to it use only TCP, will there be a problem with the number of sockets that can be open at the same time, if these clients are behind NAT? I mean that for NAT bindings to stay open, each client has to keep open TCP connection to SER all the time after registration. What is the maximum number of open sockets that Linux can handle at the same time? I've changed the maximum number of open files to 65535 (ulimit -n 65535). Doesn't this also mean the maximum number of sockets? Is this the absolute maximum? Also, does this mean the maximum number of clients that can connect to my proxy? Regards, Teemu -- Teemu Harju http://www.teemuharju.net

2 2

[Serusers] SER RTP QoS Monitoring
by Aisling O'Driscoll 02 Feb '06

02 Feb '06

Hi, I want to be able to extract QoS parameters such as jitter, delay and packet loss from VoIP calls made using my SER(both peer to peer for my public clients and also for clients where the voice will be routed with mediaproxy). I would like to develop something as opposed to purchasing a software package. >From searching the Internet the last few days I have seen that the following options seem to be available: 1) RTCP RTCP stream will contain this information. The problem is that sniffers cannot clearly distinguish RTCP packets from other UDP packets unless something like an IP address or specific UDP port is supplied. I had considered using TCPDUMP but then where do I position this in the network when voice is going peer to peer between client and I'm not sure how this can work to monitor all calls when I might not be aware of the end user IP addresses. 2) SIP INFO extension This also seem to be an option but most off the shelf phones don't seem to support this and this would required modification of a SIP ua. 3) SNMP I thought maybe SNMP might be an option but the SER snmp module no longer exists... Does anyone have any comments on the above? Are the statements that I made correct or can anyone think of other ways to monitor the voice QoS? I am trying to understand how commercial applications have accomplished this. Many thanks in advance for advice. Aisling. >---- Original Message ---- >From: clona(a)cyberhouse.no >To: ashling.odriscoll(a)cit.ie >Subject: Re: [Serusers] Status of SER SNMP module >Date: Sun, 29 Jan 2006 20:30:02 +0100 > >> >>Hi Aisling, >> >>The SNMP module is dead as a killed turtle :( >> >>-Atle >>* Aisling O'Driscoll <ashling.odriscoll(a)cit.ie> [060129 20:26]: >>> Hello, >>> >>> I'm just wondering what the current status of the SER SNMP module >is? >>> Is it currently supported and is there somewhere I can find >>> documentation on it? >>> >>> Many thanks, >>> Aisling. >>> >>> >>> >>> -------------------Legal >Disclaimer--------------------------------------- >>> >>> The above electronic mail transmission is confidential and >intended only for the person to whom it is addressed. Its contents >may be protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)iptel.org >>> http://mail.iptel.org/mailman/listinfo/serusers -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

7 7

[Serusers] high-availibility setup using f5 bigip
by Jack Wei 02 Feb '06

02 Feb '06

hi, I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do load-balancing. I'm using Asterisk as a voicemail application only and have successfully integrated SER with Asterisk without the switch. But when I try to use the switch as a load-balancer, I get lots of NAT problems. Does anyone know how to setup the switch and SER/Asterisk properly? Thanks, Jack __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

RE: [Serusers] vm.so
by Toygun Mavinil 01 Feb '06

01 Feb '06

Then where do i have to insert the codes in ser.cfg, the timeout for forward to voicemail? I tried some places but not successfull My ser.cfg is from onsip ser mediaproxy examle; Here, # $Id: nat-mediaproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Toygun >Read the more recent SEMS documentation. vm.so was deprecated. >You now use other methods (depending on whether you're using a fifo or a unix socket in SEMS). >For instance, I use: >if(!t_write_unix("/tmp/am_sock","voicemail")) { > t_reply("500","error contacting sems"); >}; On Wed, 1 Feb 2006 20:44:36 +0200, Toygun Mavinil wrote > Hi, > I installed ser 0.9.6 on fc4 > Compiled and installed with all modules but vm.so module is missing. > Looked at the source modules directory, there is no vm directory or file. > I installed sems from cvs > Sems is working > > Also > Do i have to add down lines to ser.cfg > > loadmodule "/home/srouter/sip_router/modules/vm/vm.so" > modparam("vm", "auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("vm", "email_column", "email") > modparam("vm", "subscriber_table", "subscriber") > modparam("vm", "user_column", "username") > modparam("vm", "domain_column", "domain") > > if(method=="ACK" || method=="INVITE" || method=="BYE"){ > > if (!t_newtran()) { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new transaction"); > break; > }; > > t_reply("100","Trying -- just wait a minute !"); > if(method=="INVITE"){ > log("**************** vm start - begin ******************\n"); > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the answer machine\n"); > t_reply("500","could not contact the answer machine"); > }; > log("**************** vm start - end ******************\n"); > } else if(method=="BYE"){ > log("**************** vm end - begin ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the answer machine\n"); > t_reply("500","could not contact the answer machine"); > }; > log("**************** vm end - end ******************\n"); > }; > break; > }; > if (method=="CANCEL") { > sl_send_reply("200", "cancels are junked here"); > break; > }; > sl_send_reply("501", "method not understood here"); > > > > > > > Toygun >

1 0

FW: [Serusers] vm.so
by Toygun Mavinil 01 Feb '06

01 Feb '06

Then where do i have to insert the codes in ser.cfg, the timeout for forward to voicemail? I tried some places but not successfull My ser.cfg is from onsip ser mediaproxy examle; Here, # $Id: nat-mediaproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Toygun >Read the more recent SEMS documentation. vm.so was deprecated. >You now use other methods (depending on whether you're using a fifo or a unix socket in SEMS). >For instance, I use: >if(!t_write_unix("/tmp/am_sock","voicemail")) { > t_reply("500","error contacting sems"); >}; On Wed, 1 Feb 2006 20:44:36 +0200, Toygun Mavinil wrote > Hi, > I installed ser 0.9.6 on fc4 > Compiled and installed with all modules but vm.so module is missing. > Looked at the source modules directory, there is no vm directory or file. > I installed sems from cvs > Sems is working > > Also > Do i have to add down lines to ser.cfg > > loadmodule "/home/srouter/sip_router/modules/vm/vm.so" > modparam("vm", "auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("vm", "email_column", "email") > modparam("vm", "subscriber_table", "subscriber") > modparam("vm", "user_column", "username") > modparam("vm", "domain_column", "domain") > > if(method=="ACK" || method=="INVITE" || method=="BYE"){ > > if (!t_newtran()) { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new transaction"); > break; > }; > > t_reply("100","Trying -- just wait a minute !"); > if(method=="INVITE"){ > log("**************** vm start - begin ******************\n"); > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the answer machine\n"); > t_reply("500","could not contact the answer machine"); > }; > log("**************** vm start - end ******************\n"); > } else if(method=="BYE"){ > log("**************** vm end - begin ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the answer machine\n"); > t_reply("500","could not contact the answer machine"); > }; > log("**************** vm end - end ******************\n"); > }; > break; > }; > if (method=="CANCEL") { > sl_send_reply("200", "cancels are junked here"); > break; > }; > sl_send_reply("501", "method not understood here"); > > > > > > > Toygun >

1 0

[Serusers] vm.so
by Toygun Mavinil 01 Feb '06

01 Feb '06

Hi, I installed ser 0.9.6 on fc4 Compiled and installed with all modules but vm.so module is missing. Looked at the source modules directory, there is no vm directory or file. I installed sems from cvs Sems is working Also Do i have to add down lines to ser.cfg loadmodule "/home/srouter/sip_router/modules/vm/vm.so" modparam("vm", "auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("vm", "email_column", "email") modparam("vm", "subscriber_table", "subscriber") modparam("vm", "user_column", "username") modparam("vm", "domain_column", "domain") if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); Toygun

2 1

[Users] Forward requests based on userid
by Mouncif Benziane 01 Feb '06

01 Feb '06

Hi folks, I wanted to route sip requests based on user id with it's appropriate Gateway IP address which is stored in Mysql database , is this possible? Database structure: userid,gateway 617978XXX 64.80.XXX.XXX 78189XXXX 64.100.XXX.XXXX example: if someone call 617978XXX he will be routed to it's approriate gateway at present I am using static route like: if (uri=~"^sip:617978XXX@") { rewritehost("64.80.XXX.XXX"); } if (uri=~"^sip:78189XXXX@") { rewritehost("664.100.XXX.XXXX"); } etc,.... I am afraid the config file gonna get larger if I add more than 500 users(DIDs), I am looking for an alternative to store DID numbers in database (Mysql or text) and route them accordingly with the appropriate Gateway address. Thank you so much and I will appreciate if you can help me out! Ben __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] check if transaction exists
by Andreas M. 01 Feb '06

01 Feb '06

Hi all, i want to introduce following in my configuration: CallerA calls CallerB -> ringing CallerC calls CallerB -> reject during this transaction, without that CallerB picks up the phone, CallerC should not be able to call CallerB during this transaction, After this transactions, CallerC can call CallerB also. Any ideas how to check if there is an ongoing transaction ? regards, Andreas M.

4 7

[Serusers] Re: 200 OK/SDP and ACK/SDP problem
by Carla Ares 01 Feb '06

01 Feb '06

> > Hi all, > I have a problem with audio, when the destination of a call is natted and > the other side is public. This problem is cause because the SDP info only is > present in the 200 OK and in the ACK message. > > I'm using ser (8.14) with mediaproxy (1.2.1). > > I have an H323 to SIP translator (and SIP to H323 too) connected to the > PSTN and to the SIP Proxy (SER). > H323 is the origin of the call (public client), and SIP UAC is the > destination (Natted client). > My proxy is working in state full mode and has a public IP. > > The H323 side is using slow start, so when it starts the transaction with > an INVITE, it will not send the SDP info (it will send it in the ACK > message). > The proxy forwards this message to the destination. > SIP UAC, starts ringing. When we hooks off, the SIP UAC send an 200 OK > with SDP message to the H323 side. > > This is the 200 OK / SDP that the proxy receives from the SIP side. > > No. Time Source Destination Protocol > Info > 488 28.488811 200.68.89.2 200.68.89.12 SIP/SDP > Status: 200 OK, with session description > Message body > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): 700600 6015 6015 IN IP4 > 192.168.0.101 > Owner Username: 700600 > Session ID: 6015 > Session Version: 6015 > Owner Network Type: IN > Owner Address Type: IP4 > Owner Address: 192.168.0.101 > Session Name (s): AddPac Gateway SDP > Connection Information (c): IN IP4 192.168.0.101 > Connection Network Type: IN > Connection Address Type: IP4 > Connection Address: 192.168.0.101 > Time Description, active time (t): 0 0 > Session Start Time: 0 > Session Stop Time: 0 > Media Description, name and address (m): audio 23018 RTP/AVP > 18 8 0 101 > Media Type: audio > Media Port: 23018 > Media Proto: RTP/AVP > Media Format: ITU-T G.729 > Media Format: ITU-T G.711 PCMA > Media Format: ITU-T G.711 PCMU > Media Format: 101 > > Following the ser.cfg, this 200 OK / SDP will make the proxy to use > mediaproxy (use_media_proxy()), because the destination user is natted. So > the mediaproxy module, will generate a lookup command to the > proxydispatcher.py. > > proxydispatcher[30535]: command lookup 2281401749(a)200.68.89.10 > 192.168.0.101:23018:audio 200.68.89.2 200.68.89.10 remote 200.68.89.10unknown AddPac=20SIP=20Gateway info= > from:1150316660@200.68.89.10,to:1152464490@200.68.89.10 > ,fromtag:3512844671,totag:7b07717a4 > proxydispatcher[30535]: warning: trying to lookup session with > non-existent id: '2281401749(a)200.68.89.10' > > The proxydispatcher does not recognize that command because it has not > generate the session before. So, the proxy forwards the 200 OK /SDP message > without changing the SDP info. When the H323 sides receives that info, it > thinks that it has to send RTP to the private IP. > > Then the H323 sends the ACK with SDP. > > Then, we cannot ear audio in both sides. > > Here is the problem, because, I didn't receive an INVITE with SDP that can > create the session into the dispatcher. > > > A is public, B is natted > > side A Proxy side B > > INVITE without SDP > ----------> > 100 Trying > <---------- > > INVITE without SDP > ----------> > > 100 Trying > <---------- > 180 Ringing > <---------- > 180 Ringing > <---------- > > 200 OK SDP (B private IP) > <---------- > *1* > 200 OK SDP (B private IP) > <---------- > > ACK SDP (A public IP) > ----------> > *2* > ACK SDP (mediaproxy public IP) > ----------> > > ...................................................... > RTP from B > <---------------------- > *3* > ...................................................... > > > *1* The proxy must replace the private IP of the SDP, with the ip of the > mediaproxy. The looku prequest does not work, because no session was found > *2* the proxy generates the session in the mediaproxy. So B thinks that > the RTP must go to the mediaproxy. But A never knows the mediaproxy address. > *3* RTP from A goes to anywhere, because A does not know where it is the > private adrress of B. > > Do you know a solution to this? Is the sequence of the message all right? Regards. Carla

2 5

[Serusers] Here I go again...Call type processing Section problem?
by Juan 01 Feb '06

01 Feb '06

Hi. As I had no answers to this question, here it is again. Till now I was happy using SER handling communications between UAs using nathelper and RPT proxy. Now I changed the ser.cfg script to the one with PSTN support and none of the UAs get registered. I am getting a 403 Forbidden. Analyzing the script I understood that this 403 was coming from the Call Type Processing Section so I changed if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { and so on to if (uri!=myself) and now the UAs are registering again. But of course I can't make calls. So I would like to know how this< is_uri_host_local and is_from _local> work and what should I do to make them allow my UAs to register. I am attaching the ser.cfg that doesn't work. Thanks in advance Juan Ferrari

2 2

[Users] TM bug?
by Greg Fausak 01 Feb '06

01 Feb '06

I am running a proxy pushing 2 million calls a day. Every now and then, I drop 5 or 10 BYE messages. I've attached the sip_scenario log file, F16,F17,F20 all send BYE messages to 12.46.104.62. F21 is sending the BYE form 12.46.104.62 to 12.46.104.252. The OK comes back, but is never relayed. BYEs continue to retry until 12.46.104.62 finally gives up and sends a Request timeout 408 in frame F38. So, F38 is evidence that the transaction started. Right? Why are all of the BYEs ignored, and why isn't the OK relayed? I have xlog output from 12.46.104.58 indicating each time a BYE is loose-routed to 12.46.104.62. The syslog file shows each attempt (F15,F18,F19,F23, etc). I also have xlog output from 12.46.104.62, however, the syslog on that machine never shows receiving or loose-routing a BYE with this callid. It seems like receiving more than one BYE in a short period of time is screwing up the t_relay()??? The calls are being generated with sipp, and 999,990 times out of a million it works. ---greg -- Greg Fausak greg(a)thursday.com

2 2

[Serusers] [VOIPSEC] any information regarding adding DTLS using OpenSSL
by Pjothi 01 Feb '06

01 Feb '06

Dear all, I am a student and am trying to setup a demonstration with TLS and DTLS support between a SIP client and a Proxy. Has anyone some information regarding adding DTLS support for a SIP client/Proxy or a more generic one using OpenSSL. I just need it for demonstration purposes and so error handling's if any need not be complete. Just a simple support and a demonstration of both would do fine. If anyone has any information regarding this, may be a how to start or any introductory article would be of great help, kindly let me know. This would be of great help. Any suggestions regarding the same are most welcome. kindly let me know and thank you very much. regards, Pjothi _______________________________________________ Voipsec mailing list Voipsec(a)voipsa.org http://voipsa.org/mailman/listinfo/voipsec_voipsa.org --9B095B5ADSN=_01C62670194DF0AE000000EForinoco-- ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email ______________________________________________________________________

1 0

Re: [Serusers] failover and VRRP
by hgaillac-sip＠yahoo.fr 01 Feb '06

01 Feb '06

> I use NAGIOS to > monitor SER to know when a server of applications on > that server have ... Can you tell me more ? Harry --- Steve Blair <blairs(a)isc.upenn.edu> a écrit : > > > hgaillac-sip(a)yahoo.fr wrote: > > >Hello, > > > >Ok SRV records point to proxy sip server but if > this > >one is down before or during connection !!! > > > > > > > During a connection doesn't matter in my > environment. I am running SER > in a stateless > mode. Once the call is setup media flows between > endpoints and the proxy > is out of the > connection. I am not using NAT either. > > > SER1 > >DNS servers=== [?] =====sip agents > > SER2 > > > >How [?] can it provide H-A (heartbeat soft) between > >ser boxes. > > > > > > > Neither the SRV or VRRP can provide a heartbeat > between SER boxes. You'd > need an application running on the SER boxes to do > that. I rely upon > t_replicate to update the backup during > registration, I use NAGIOS to > monitor SER to know when a server of applications on > that server have > failed and I use an SRV record defined in each phone > to identify the > proxy. This way the phone will failover on it's own > during call setup. > Of course this doesn't help during a call but as I > said I am running SER > in a stateless mode so the proxy "steps out" once > the call is setup. > > >If one of these sip proxy failed during a call can > we > >redirect the call to the other proxy ? > > > > > I'm not sure how you would accomplish this. > > >Regards > >Harry > > > >--- Steve Blair <blairs(a)isc.upenn.edu> a écrit : > > > > > > > >>Sorry. I don't do diagrams :-) Which piece of this > >>explanation is unclear? > >> > >>-Steve > >> > >>hgaillac-sip(a)yahoo.fr wrote: > >> > >> > >> > >>>Could you please send me a diagram ? > >>> > >>>Harry > >>>--- Steve Blair <blairs(a)isc.upenn.edu> a écrit : > >>> > >>> > >>> > >>> > >>> > >>>>VRRP for layer 3. H-A woud depend upon your > >>>>availability requirements. > >>>>In our case we use an SRV record to identify > >>>>multiple proxy servers each > >>>>with their own priority and weight. Phones > >>>> > >>>> > >>register > >> > >> > >>>>using the SRV name > >>>>as the proxy address. Each proxy uses > t_replicate > >>>> > >>>> > >>to > >> > >> > >>>>replicate > >>>>registration data. So far this works for us. > >>>> > >>>>_Steve > >>>> > >>>>hgaillac-sip(a)yahoo.fr wrote: > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>>Ok so vrrp for layer 2/3 but which solution for > >>>>> > >>>>> > >>H-A > >> > >> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>? > >>>> > >>>> > >>>> > >>>> > >>>>>Harry > >>>>>--- Steve Blair <blairs(a)isc.upenn.edu> a écrit > : > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>Harry: > >>>>>> > >>>>>> Assuming you are talking about running VRRP > in > >>>>>>your routers and not > >>>>>>on the Asterisk host itself then the answer is > a > >>>>>>qualified yes. VRRP by > >>>>>>itself won't give you the transparency of an > SRV > >>>>>>record for identifying > >>>>>>your proxy servers. It will allow you to have > >>>>>>automatic failover to > >>>>>>backup gateways should the primary gateway > >>>>>> > >>>>>> > >>to/from > >> > >> > >>>>>>your "SER subnet" fail. > >>>>>> > >>>>>>-Steve > >>>>>> > >>>>>>hgaillac-sip(a)yahoo.fr wrote: > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>>>hello, > >>>>>>> > >>>>>>>Can we use VRRP when a SER is down ? > >>>>>>> > >>>>>>> > >>>>>>> SER1 (master) > >>>>>>>sip agents == || === Asterisk Farm==sip/pstn > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>gateway > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>>> SER2 (slave) > >>>>>>> > >>>>>>>Regards > >>>>>>>Harry > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>___________________________________________________________________________ > >>>> > === message truncated === ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

3 3

[Serusers] SIP message bodies sometimes corrupted
by Jon Burke 01 Feb '06

01 Feb '06

Hello, I'm having trouble with SER (0.9.4) sometimes corrupting the bodies of SIP messages that it forwards. I'm using binary message bodies, with appropriate Accept, Content-Type and Content-Length headers which the proxy server should forward without modification. Sometimes the messages are forwarded correctly with all headers and the message body intact. Other times, one byte has gone missing from the end of the message body, so that the value in the Content-Length header is one more than the actual number of bytes in the message body. The receiving UA then rejects this message because it is malformed. From repeated testing, it looks as though this problem happens whenever the number of bytes in the SIP message (going out from SER) prior to the body is an even number. When there are an odd number of bytes, the message body is always uncorrupted. Anybody have any ideas what's happening here or how to fix it? Jon.

2 1

[Serusers] is SER UDP capable ?
by Pjothi 01 Feb '06

01 Feb '06

Dear all, Is SER proxy udp capabale, can I use an UDP client application to communicate with SER ? thanks and regards, Pjothi

2 1

[Users] pdt module enhanced with multidomain support
by Elena Ramona Modroiu 01 Feb '06

01 Feb '06

Hello all, the pdt module was enhanced to support multidomain translation of the prefixes, i.e. a new R-URI domain is searched by prefix and domain of FromURI. Until now the search was performed by prefix only. ************** This means the database table will have three columns (sdomain, prefix, domain), where sdomain stands for a source domain to which a set of (prefixes,domain) pairs could be assinged. Database creation example: DROP DATABASE IF EXISTS pdt; CREATE DATABASE pdt; USE pdt; -- create table CREATE TABLE pd_multidomain ( sdomain VARCHAR(255) NOT NULL, prefix VARCHAR(32) NOT NULL, domain VARCHAR(255) NOT NULL DEFAULT "", PRIMARY KEY (sdomain, prefix) ); ************** The module exports one function prefix2domain(mode) to be used from script to update the RURI if this is in a PSTN format. For details about the 'mode' parameter please refer to the README file. The module parameter called 'prefix' is available to define which URIs must be translated. Here it is an example how to set it in the script file modparam("pdt", "prefix", "00"). This means only the RURIs in the format 00xxxxx@yyyyy will be updated if predix2domain() is called in the script. By default is set to NULL, and all RURIs will be updated if prefix2domain() is called. An example of translation: -- Input: modparam("pdt", "prefix", "00") Database entries (sdomain - prefix - domain): (siphub.org, 123, alpha.org) (siphub.org, 124, beta.org) (siphub.org, 126, gamma.org) (openser.org, 123, one.org) (openser.org, 124, two.org) (openser.org, 125, three.org) Message: RURI: 00123456789(a)mydomain.com FromURI: alice(a)siphub.org -- Translation via prefix2domain(0) -- Output RURI: 456789(a)alpha.org ************** Three fifo commands are available for adding, deleting and listing the (sdomain, prefix, domain) entries in the database. /* * How to add a new entry - fifo command example * * --- * :pdt_add:[response_file]\n * sdomain\n * prefix\n * domain\n * \n * -- * * - '.' (dot) means NULL value and is not allowed */ /* * How to delete existent entries - fifo command example: * * --- * :pdt_delete:[response_file]\n * sdomain * domain\n * \n * -- * * - the specified (sdomain, *, domain) entry will be deleted * - '.' (dot) means NULL value and is not allowed */ /* * How to see already defined entries - fifo command example: * * --- * :pdt_list:[response_file]\n * sdomain\n * prefix\n * domain\n * \n * -- * * - '.' (dot) means NULL value * - the comparison operation is 'START WITH' -- if domain is 'a' then * all domains starting with 'a' are listed */ In the module directory under ./doc/ there are available example shell scripts for these commands. Any feedback is highly appreciated. Regards, Ramona

1 0

RE: [Serusers] Serweb does not send mail on registration
by Bart J. Smit 01 Feb '06

01 Feb '06

Hi Karel, Fixed! 8-) I did 'yum -y install php-imap' which fixed the imap_8bit() function call. I then got "sorry -- cannot open write fifo" on the registration confirmation page. This file is not writable by my httpd user (user apache on CentOS) so I set the ownership to root.apache. This is reset to root.root when ser starts, so I added 'chown root.apache /tmp/ser_fifo' to the start section of the /etc/init.d/ser script, just before the 'return $RETVAL' line. Serweb now works a treat. Thanks again for your code and your help, Bart... -----Original Message----- From: Karel Kozlik [mailto:karel@iptel.org] Sent: 01 February 2006 10:30 To: Bart J. Smit Cc: SER Users Subject: Re: [Serusers] Serweb does not send mail on registration well, you need the imap extension for php. It should be sufficient to add this line to your php.ini: extension=imap.so Karel Bart J. Smit napsal(a): > Thanks Karel, > > Good thinking; I forgot to check the apache logs. > > The error log gave a vital clue: > > PHP Fatal error: Call to undefined function: imap_8bit() in > /var/www/serweb-0.9.4/html/functions.php on line 310. > > I'll go and check the PHP to see if I need additional PEAR modules or a > patch to the script. > > Bart... > > -----Original Message----- > From: Karel Kozlik [mailto:karel@iptel.org] > Sent: 01 February 2006 07:55 > To: Bart J. Smit > Cc: SER Users > Subject: Re: [Serusers] Serweb does not send mail on registration > > Hello, > is there an error message in PHP log. Or in apache error log? > > Karel > > Bart J. Smit napsal(a): > >>I have serweb-0.9.4 installed on CentOS 4.2 with MySQL 4.1.12-3 and >>ser-0.9.6-1 compiled from the source RPM at >>http://ftp.upjs.sk/pub/users/sal/Fedora/4/voip. When I register a new >>user, the PHP script goes to an empty page without title and the email >>never gets sent to the address entered on the registration form. >> >>The database gets the correct user details inserted into the pending >>table. The sendnotification and greeting fields remain empty. >> >>I have modified the line in config_data_layer.php for the collation >>character set to read: >>$config->data_sql->collation = "latin1_swedish_ci"; >>This got rid of a DB error in serweb which I traced through pear >>logging: >>Illegal mix of collations (latin1_swedish_ci,IMPLICIT) and >>(utf8_general_ci,COERCIBLE) for operation '='] >> >>I have sent a test email with the debugging/mail.php script and it >>arrives fine. I have tested the registration page with local > > recipients, > >>recipients on the LAN and external recipients but the behaviour stays >>the same. >> >>All the pages I have googled on serweb registration problems always >>state that the emails are sent out correctly, so I'm likely missing >>something obvious. >> >>Can anybody help me get the registration to work? >> >>Thanks, >> >>Bart... >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers

1 0

RE: [Serusers] Serweb does not send mail on registration
by Bart J. Smit 01 Feb '06

01 Feb '06

Thanks Karel, Good thinking; I forgot to check the apache logs. The error log gave a vital clue: PHP Fatal error: Call to undefined function: imap_8bit() in /var/www/serweb-0.9.4/html/functions.php on line 310. I'll go and check the PHP to see if I need additional PEAR modules or a patch to the script. Bart... -----Original Message----- From: Karel Kozlik [mailto:karel@iptel.org] Sent: 01 February 2006 07:55 To: Bart J. Smit Cc: SER Users Subject: Re: [Serusers] Serweb does not send mail on registration Hello, is there an error message in PHP log. Or in apache error log? Karel Bart J. Smit napsal(a): > I have serweb-0.9.4 installed on CentOS 4.2 with MySQL 4.1.12-3 and > ser-0.9.6-1 compiled from the source RPM at > http://ftp.upjs.sk/pub/users/sal/Fedora/4/voip. When I register a new > user, the PHP script goes to an empty page without title and the email > never gets sent to the address entered on the registration form. > > The database gets the correct user details inserted into the pending > table. The sendnotification and greeting fields remain empty. > > I have modified the line in config_data_layer.php for the collation > character set to read: > $config->data_sql->collation = "latin1_swedish_ci"; > This got rid of a DB error in serweb which I traced through pear > logging: > Illegal mix of collations (latin1_swedish_ci,IMPLICIT) and > (utf8_general_ci,COERCIBLE) for operation '='] > > I have sent a test email with the debugging/mail.php script and it > arrives fine. I have tested the registration page with local recipients, > recipients on the LAN and external recipients but the behaviour stays > the same. > > All the pages I have googled on serweb registration problems always > state that the emails are sent out correctly, so I'm likely missing > something obvious. > > Can anybody help me get the registration to work? > > Thanks, > > Bart... > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

2 1

Re: [Fwd: [Users] enum patch]
by Greg Fausak 01 Feb '06

01 Feb '06

Juha, I've updated the documentation. I don't know how to produce output, so I don't know if it works. I haven't done a patch before either, so, let me know if this is the preferred format or if you want to see it a different way. By the way, when I was reading the documentation I noticed that the enum query puts the highest priority query in the ruri, then the rest of them are stored in new branches with a q value. I haven't actually tried this. Is it the plan that these branches will be looped through serially upon failure without having a retry route block? The new patch is attached. Thanks, -g On Jan 29, 2006, at 4:27 AM, Bogdan-Andrei Iancu wrote: > To be more precisely, the file is modules/enum/doc/enum_user.sgml > > bogdan > > Juha Heinanen wrote: > >>> is the documentation the README file? >>> >> >> you have to edit the xml file from which README is generated >> automatically. >> >> -- juha >> >> >

4 3

[Users] dispatcher module
by Jayesh Nambiar 01 Feb '06

01 Feb '06

Hi all, I am trying to use the dispatcher module. I have two gateway address in the dispatcher.list file. But when it reaches that particular gateway, it does not rewrite the host with that gateway. Is dispatcher not supposed to do it?? One more thing.... when I used the function forward(); for forwarding the call to set of gateways, the openser shows an error saying that it is "bad forwardarguement". openser needs the ip addr and the port in the forward function, but then how will it look at the destination set. It worked when I used t_relay(), but I need to forward it statelessly. I just used the function as it was shown in the example of the dispatcher module in the docs.. can someone please help me on this.. Thanks in advance. jayesh. --------------------------------- Jiyo cricket on Yahoo! India cricket

2 1

[Users] Using uac_replace_from to munge username between ser and Asterisk?
by Barry Flanagan 01 Feb '06

01 Feb '06

Hi, I am trying to get a multi-domain setup working between OpenSER and Asterisk (PSTN gateway and Voicemail). It appears that I need to munge the username before it goes to Asterisk so that it includes the domain element, otherwise there is clashing in the username namespace on Asterisk. It looks like I should be able to use uac_replace_from to change the From URI from user(a)domain.com to user_domain(a)domain.com so that then I can set up the users in Asterisk as [user_domain] Does this sound sane, and could someone tell me how I can accomplish this in OpenSER? I have tried a few things but they all end up as bad headers. Thanks in advance. -- -Barry Flanagan

4 6

Re: [Serusers] [serweb]Problem with SerWeb
by mkumar＠mantragroup.com 01 Feb '06

01 Feb '06

Hi All, I tried installing 0.9.2 and 0.9.4 but both behave in the same way, only if I hit 5 to 10 times on login then one can login and in admin only ACl changes and user preferences are working. Server monitoring and admin privileges are not working. Things which are working will work only if I hit 5 to 10 times on them only. Please help me, I do not know where I am going wrong, I followed all the steps in serweb INSTALL doc. Thanks, Manoj. Quoting Karel Kozlik <karel(a)iptel.org>: > Hi, > I looked at your web server but it's behaviour is very strange. It > sometimes don't reply, sometimes not return whole html page. For > example the page > http://www.mantragroup.com/serweb/html/admin/users.php is often cut > off in the middle. > > Try examine your logs. Access and error log of apache and log of php. > > Karel > > mkumar(a)mantragroup.com napsal(a): >> Hi Karel Kozlik, >> >> Thanks for the reply. Initially I tried using serweb admin mode but >> for me only >> the initial page http://www.mantragroup.com/serweb/html/admin/users.php is >> displayed from there if I hit on admin privileges or server >> monitoring or user >> preferences nothing opens, it still shows the same initial page of >> admin login. >> >> Thanks, >> Manoj. >> >> Quoting Karel Kozlik <karel(a)iptel.org>: >> >>> Hi, >>> please (CC) the forum in your replies in order to others can read >>> the answers. >>> >>> To your problem: >>> the correct values to allow change ACL are: >>> >>> priv_name: acl_control >>> priv_value: int >>> >>> priv_name: acl_control >>> priv_value: ld >>> >>> priv_name: acl_control >>> priv_value: local >>> >>> But you realy do not need to know this! >>> >>> Instead of changeing values in DB you can simply login to serweb as admin: >>> - click to tab 'admin privileges' >>> - click to link 'changes privilges' for user 'admin' >>> - mark the checkboxes in 'ACL control' area >>> - click to 'save' >>> >>> Now you should be able to change ACL of other users. >>> >>> Karel >>> >>> >>> >>> mkumar(a)mantragroup.com napsal(a): >>> >>>> Hi Karel Kozlik, >>>> >>>> I really do not know what are the correct values that must be given under >>>> priv_name, for admin I gave is_admin, change_privileges and for giving acl >>>> permissions I tried giving local,ld,acl_ld,acl_local but none worked and I >>>> still get "You haven't any privileges to control ACL" when I log >>>> in to admin, >>>> can you please give me syntax which grants acl change privilges to >>>> admin. I >>>> tried searching on net and doc which came with installation but >>>> none helped me. >>>> Please spare a minute and help me configure serweb. >>>> >>>> Thanks, >>>> Manoj. >>>> >>>> Quoting Karel Kozlik <karel(a)iptel.org>: >>>> >>>>> Hi, >>>>> that is OK. You can have more admins in serweb. And you can >>>>> define privileges for changeing ACL for each admin. To do that, >>>>> click to tab 'admin privileges' and select 'change privileges' >>>>> for some admin. Here you may define which values of ACL this >>>>> admin may change. >>>>> >>>>> Karel >>>>> >>>>> mkumar(a)mantragroup.com napsal(a): >>>>> >>>>>> Hi All, >>>>>> >>>>>> I was able to install serweb and login to it both from admin and >>>>>> user end. But >>>>>> in admin end I have a problem, when I hit on ACL of any user >>>>>> that appered when >>>>>> I first login as admin, I get a error "You haven't any >>>>>> privileges to control >>>>>> ACL". In the DataBase I gave the following privileges to admin >>>>>> >>>>>> username admin >>>>>> domain mantragroup.com >>>>>> priv_name change_privileges >>>>>> priv_value 1 >>>>>> >>>>>> username admin >>>>>> domain mantragroup.com >>>>>> priv_name change_privileges >>>>>> priv_value 1 >>>>>> >>>>>> Someone please help me and guide me to inderstand where I am >>>>>> going wrong. >>>>>> >>>>>> Thanks, >>>>>> Manoj >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Serusers mailing list >>>>>> serusers(a)lists.iptel.org >>>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>> >>>>> >>>>> >>>> >>>> >>>> >>> >> >> >> >

2 2

[Serusers] About Serweb installation
by Guty 01 Feb '06

01 Feb '06

> I'm running my SER on a debian 3.1. > I've installed (i think so) all necesary stuff to run SER whit mysql > support and serweb interface. > when I try to acces it, i receive the following message: > > Warning: main(DB.php): failed to open stream: No such file or directory in > /home/gustavo/serweb/html/main_prepend.php on line 38 > > Fatal error: main(): Failed opening required 'DB.php' > (include_path='.:/usr/share/php:/usr/share/pear') in > /home/gustavo/serweb/html/main_prepend.php on line 38 > > > I'll preciate if anyone could help me.... > (as you could see, I'm new to all this stuff... ) > Thank you.....

2 1

[Serusers] Presence + XCAP root problem with multiple domains
by Adrian Georgescu 01 Feb '06

01 Feb '06

Hello Vaclav, I cannot manage to handle the same username under different domain names using the same XCAP tree. Data location for user1(a)example1.com will conflict with data location for user1(a)example2.com on the filesystem is the same: /xcap-root/pres-rules/users/user1/presence-rules.xml My XCAP server can generate separate trees for each domain and create the XCAP structure under each domain (like /xcap-root/example1/pres-rules/users/ ) but in this case SER should be able to build a dynamic root like: https://xcap.example.com/$domain$/ In Eyebeam I can configure the xcap-root statically but obviously if I do this for multiple domains, SER should be able to build it from information available in the PUBLISH/SUBSCRIBE request so that it can locate the right tree on the XCAP server. Do you see any other solution for this problem? Regards, Adrian

2 1

[Serusers] Serweb does not send mail on registration
by Bart J. Smit 01 Feb '06

01 Feb '06

I have serweb-0.9.4 installed on CentOS 4.2 with MySQL 4.1.12-3 and ser-0.9.6-1 compiled from the source RPM at http://ftp.upjs.sk/pub/users/sal/Fedora/4/voip. When I register a new user, the PHP script goes to an empty page without title and the email never gets sent to the address entered on the registration form. The database gets the correct user details inserted into the pending table. The sendnotification and greeting fields remain empty. I have modified the line in config_data_layer.php for the collation character set to read: $config->data_sql->collation = "latin1_swedish_ci"; This got rid of a DB error in serweb which I traced through pear logging: Illegal mix of collations (latin1_swedish_ci,IMPLICIT) and (utf8_general_ci,COERCIBLE) for operation '='] I have sent a test email with the debugging/mail.php script and it arrives fine. I have tested the registration page with local recipients, recipients on the LAN and external recipients but the behaviour stays the same. All the pages I have googled on serweb registration problems always state that the emails are sent out correctly, so I'm likely missing something obvious. Can anybody help me get the registration to work? Thanks, Bart...

2 1

[Serusers] windows messenger 5.1 and IM
by Yi Zheng 01 Feb '06

01 Feb '06

Note: forwarded message attached.

2 1

[Serusers] [serweb] Problem with admin privileges, account, accounting, aliases
by mkumar＠mantragroup.com 01 Feb '06

01 Feb '06

Hi All, I installed serweb and able to login both as admin and user. But some features are not at all working for me and also I could not see any error message. The following tabs in admin mode are not working account,accounting,aliases in users admin privileges server monitoring after enabling cron jobs The above tabs common to user mode like accounting are also not working. I have mysql version 3.23.58. I am getting these problems only because mysql version is <4.0 or should I update or add anything to my ser.cfg or any other place? Someone please help me. I do not know where I am going wrong or what to do and there are no errors as well. Thanks, Manoj.

1 0

[Serusers] SER and Cisco ATA 186
by R2 01 Feb '06

01 Feb '06

Hi all, How I can supress of uri user=phone ? This ATA receive call from PSTN, but, when I send calls to PSTN, i don't obtain response. The integration to PSTN is on Asterisk. Thanks R2

2 1

[Serusers] Help on SER with Mediaproxy
by Muhammad Asif Ali 01 Feb '06

01 Feb '06

Hi All I have a scenario in which I want to deploy SER with mediaproxy & allow everyone/anyone connecting with my SER to talk to each other. I am using ONSIP gettingStarted document. In order to allow everyone to connect to my SER without authentication I commented the following portion in ser.cfg: in Route[2] #if (!www_authorize("","subscriber")) { # www_challenge("","0"); # break; #}; #if (!check_to()) { # sl_send_reply("401", "Unauthorized"); # break; #}; # consume_credentials(); and in Route[3] # if (!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # break; #} else if (!check_from()) { # sl_send_reply("403", "Use From=ID"); # break; #}; #consume_credentials(); Now when I try to connect 2 UA (Simpleopal) from behind a same NAT, they do get registered, & records also appear in location table of DataBase. But when I place call from UA1 to UA2 like "call sip:UA2@aa.bb.cc.dd", where aa.bb.cc.dd is the public IP of my SER machine (mediaproxy is also running on same machine), it ends up showing transport error. On SER side: A) SER logs say: on lookup() "UA2 not found in usrloc" and on is_local() "Realm 'ab.cc.dd.ee' is not local". the is_lcoal appears again saying "Realm 192.168.0.241 is not local" where 192.168.0.241 is UA2's local IP. B)Mediaproxy log says: command request c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1 203.82.51.9:5000:audio 203.82.51.9 aa.bb.cc.dd remote 192.168.0.241 remote OPAL/2.0 info= from:UA1@aa.bb.cc.dd,to:UA2@aa.bb.cc.dd ,fromtag:b09168c6-fbf4-1810-80b4-0050badb4a03,totag: session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: started. listening on aa.bb.cc.dd:35000 command execution time: 9.35 ms command request c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1 203.82.51.9:5000:audio 203.82.51.9 aa.bb.cc.dd remote 192.168.0.241 remote OPAL/2.0 info= from:UA1@aa.bb.cc.dd,to:UA2@aa.bb.cc.dd ,fromtag:b09168c6-fbf4-1810-80b4-0050badb4a03,totag: command execution time: 0.60 ms session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: ended (did timeout). (203.82.51.9) is my LAN's public IP aa.bb.cc.dd is the public IP of SER machine C) using ngrep I found out that it sends an "SIP/2.0 408 Request Timeout" cant figure out whats going on, anybody having any idea please help. Thanx in advance Regards Muhammad Asif Ali

2 4

[Serusers] any info reg adding DTLS using OpenSSL
by Pjothi 01 Feb '06

01 Feb '06

Dear all, I am a student and am trying to setup a demonstration with TLS and DTLS support between a SIP client and a Proxy. Has anyone some information regarding adding DTLS support for a SIP client/Proxy or a more generic one using OpenSSL. I just need it for demonstration purposes and so error handling's if any need not be complete. Just a simple support and a demonstration of both would do fine. If anyone has any information regarding this, may be a how to start or any introductory article would be of great help, kindly let me know. This would be of great help. Any suggestions regarding the same are most welcome. kindly let me know and thank you very much. regards, Pjothi

2 1

[Serusers] STUN and symmetric nat again
by Istvan De 01 Feb '06

01 Feb '06

Hi! I've read about the above topic in the arcives and now I've got the same problem. User A is behind a symmetric NAT -> a new port is bound for every different (external ip, external port) pair. User B has a connection to the net without any NAT or firewall. User A wants to have a SIP+RTP session with user B. How? I guess basically what is needed is forming of a SIP message like "send the your reply to the same ip and same port from where you have received this" and user A can send this to user B. For the RTP session, a pretty similar thing would be needed like, A says to B: "I send you the RTP stream on your x port, send your rtp stream back to the same ip and port from where you receive mine". How to setup such a SIP packet? Or maybe there is a standard solution for this already? (STUN wont help in this case I guess, and TURN is not an option for me.) I've read about "symmetric SIP + symmetric RTP" maybe that is the same as I've described above? Thanks Istvan

3 2

[Serusers] adress book
by Joao Pereira 01 Feb '06

01 Feb '06

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

4 4

Re: [Serusers] Monitoring
by hgaillac-sip＠yahoo.fr 01 Feb '06

01 Feb '06

Do you Know HSRP from Cisco System ? http://www.cisco.com/univercd/cc/td/doc/cisintwk/ics/cs009.htm Harry --- Hendrik Scholz <hendrik.scholz(a)freenet-ag.de> a écrit : > Hi! > > hgaillac-sip(a)yahoo.fr wrote: > > What about VRRP for failover ? > > > > SER1 (master) > > sip agents == || === Asterisk Farm==sip/pstn > gateway > > SER2 (slave) > > What do you mean by that? > > Somewhat confused, > Hendrik > > -- > freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 > Kiel, Germany > Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 > Internet: http://www.freenet.de, eMail: > hendrik.scholz(a)freenet-ag.de > ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

2 1

[Serusers] About Serweb installation
by Gustavo Moreira 01 Feb '06

01 Feb '06

> I'm running my SER on a debian 3.1. > I've installed (i think so) all necesary stuff to run SER whit mysql > support and serweb interface. > when I try to acces it, i receive the following message: > > Warning: main(DB.php): failed to open stream: No such file or directory in > /home/gustavo/serweb/html/main_prepend.php on line 38 > > Fatal error: main(): Failed opening required 'DB.php' > (include_path='.:/usr/share/php:/usr/share/pear') in > /home/gustavo/serweb/html/main_prepend.php on line 38 > > > I'll preciate if anyone could help me.... > (as you could see, I'm new to all this stuff... ) > Thank you.....

1 0

[Serusers] About Serweb installation
by Guty 01 Feb '06

01 Feb '06

I'm running my SER on a debian 3.1. I've installed (i think so) all necesary stuff to run SER whit mysql support and serweb interface. when I try to acces it, i receive the following message: Warning: main(DB.php): failed to open stream: No such file or directory in /home/gustavo/serweb/html/main_prepend.php on line 38 Fatal error: main(): Failed opening required 'DB.php' (include_path='.:/usr/share/php:/usr/share/pear') in /home/gustavo/serweb/html/main_prepend.php on line 38 I'll preciate if anyone could help me.... (as you could see, I'm new to all this stuff... ) Thank you.....

1 0

[Serusers] windows messenger 5.1 and IM
by Yi Zheng 31 Jan '06

31 Jan '06

I have two window messenger 5.1 clients on the same subnet as the SER server w/ pa module. They subscribe/notify to each other ok. HOwever they cannot IM each other. The callee, after receiveing INVITE, sent back a 100 trying, then 200 OK, and then immediately BYE. I searched thru archives and looks like this is a known issue with wm. What is the solution or workaround to this? Here is the messages I captured: U 10.1.104.251:1172 -> 10.1.104.254:5060 INVITE sip:104pc1@10.1.104.254 SIP/2.0..Via: SIP/2.0/UDP 10.1.104.251:8053 ..Max-Forwards: 70..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>; tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1 .104.254>..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Cont act: <sip:10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Type: applicati on/sdp..Content-Length: 119....v=0..o=- 0 0 IN IP4 10.1.104.251..s=session ..c=IN IP4 10.1.104.251..t=0 0..m=message 5060 sip sip:104pc2@10.1.104.254 .. # U 10.1.104.254:5060 -> 10.1.104.251:8053 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.1. 104.251:8053..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>;tag=d4 6c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1.104.2 54>..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Server: Si p EXpress router (0.10.99-dev30-tm-timers-pa-3 (i386/linux))..Content-Leng th: 0..Warning: 392 10.1.104.254:5060 "Noisy feedback tells: pid=23866 re q_src_ip=10.1.104.251 req_src_port=1172 in_uri=sip:104pc1@10.1.104.254 out _uri=sip:10.1.104.248:13018 via_cnt==1".... # U 10.1.104.254:5060 -> 10.1.104.248:13018 INVITE sip:10.1.104.248:13018 SIP/2.0..Record-Route: <sip:10.1.104.254;fta g=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on>..Via: SIP/2.0/UDP 10.1.104.254;b ranch=z9hG4bK2216.fea17c01.0..Via: SIP/2.0/UDP 10.1.104.251:8053..Max-Forw ards: 16..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>;tag=d46c61 cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1.104.254>. .Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Contact: <sip: 10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Type: application/sdp..Co ntent-Length: 119..P-hint: usrloc applied....v=0..o=- 0 0 IN IP4 10.1.104. 251..s=session..c=IN IP4 10.1.104.251..t=0 0..m=message 5060 sip sip:104pc 2(a)10.1.104.254.. # U 10.1.104.248:1334 -> 10.1.104.254:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.1.104.254;branch=z9hG4bK2216.fea17 c01.0..Via: SIP/2.0/UDP 10.1.104.251:8053..From: "104pc2(a)10.1.104.254" <si p:104pc2@10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d136 2..To: <sip:104pc1@10.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..Cal l-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..User-Agent: RTC/1. 3..Content-Length: 0.... # U 10.1.104.248:1334 -> 10.1.104.254:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.1.104.254;branch=z9hG4bK2216.fea17c01. 0..Via: SIP/2.0/UDP 10.1.104.251:8053..From: "104pc2(a)10.1.104.254" <sip:10 ;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..T">4pc2(a)10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..T o: <sip:104pc1@10.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..Call-ID : 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Record-Route: <sip:10. 1.104.254;ftag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on>..Contact: <sip:10.1 .104.248:13018>..User-Agent: RTC/1.3..Content-Type: application/sdp..Conte nt-Length: 119....v=0..o=- 0 0 IN IP4 10.1.104.248..s=session..c=IN IP4 10 .1.104.248..t=0 0..m=message 5060 sip sip:104pc1@10.1.104.254.. # U 10.1.104.248:1334 -> 10.1.104.254:5060 BYE sip:10.1.104.254;ftag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on SIP/2.0.. Via: SIP/2.0/UDP 10.1.104.248:13018..Max-Forwards: 70..From: <sip:104pc1@1 0.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..To: "104pc2(a)10.1.104.25 4" <sip:104pc2@10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=692 00d1362..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 BYE..Route: <s ip:10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Length: 0.... ##

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[Users] Parsing SIP responses
by Ovidiu Sas 31 Jan '06

31 Jan '06

Hi, I configured openser to handle redirect responses and re-route calls based on the Contact header from the 3xx response using the uac_redirect module. In addition to that, I would like to transfer some headers from the 3xx response into the redirected INVITE (see diagram): UAC openser UAS1 UAS2 | | | | |---INVITE--->| | | | |---INVITE--->| | | |<--3XX-------| | | | with Diversion header | | |---ACK------>| | | |---------------INVITE------->| | | | with Diversion header | | | from 3xx response from UAS1 Is there an API that will allow me to search for headers inside a response and save them in to an AVP? Thx, -ovi

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Re: [Serusers] no SRV query after rewriting R-URI?
by alan chiu 31 Jan '06

31 Jan '06

great, thanks for the clarification...so simply, rewrite("host") would trigger SRV lookup. --- Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > Hi Alan! > > That's the correct behaviour. Please read RFC 3263. > If the URI contains a > port, e.g. inside.acme.com:5060 there must not be an > SRV lookup. Because > the SRV lookup leads to a port, which is already > defined. > > regards > klaus > > On Tue, January 31, 2006 20:28, alan chiu said: > > Server: Sip EXpress router (0.8.12 > (i86pc/solaris)) > > > > when endpoint sends 18005551212(a)sip.acme.com, SER > > (without rewriting of R-URI, looks it up with SRV > > *************** > > > > 386.656875 192.168.1.246 -> 10.100.0.111 SIP/SDP > > Request: INVITE sip:18005551212@sip.acme.com, with > > session description > > 386.657307 10.100.0.111 -> 192.168.1.246 SIP > Status: > > 100 trying -- your call is important to us > > 88.811658 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query SRV _sip._udp.sip.acme.com > > 98.825208 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response, No such name > > 98.825496 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query A sip.acme.com > > 108.845136 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response, No such name > > 108.845468 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query AAAA sip.acme.com > > 118.865215 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response, No such name > > 416.711408 10.100.0.111 -> 192.168.1.246 SIP > Status: > > 478 Unresolveable destination (478/TM) > > 407.921364 192.168.1.246 -> 10.100.0.111 SIP > Request: > > ACK sip:18005551212@sip.acme.com > > > > > *********************************************************************************** > > > > > > ************* > > after rewritehostport("inside.acme.com:5060"), SER > > only does an A lookup > > *************** > > > > 424.851557 192.168.1.246 -> 10.100.0.111 SIP/SDP > > Request: INVITE sip:18005551212@nat.foo.com, with > > session descripti > > on > > 415.971953 10.100.0.111 -> 192.168.1.246 SIP > Status: > > 100 trying -- your call is important to us > > 127.006508 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query A inside.foo.com > > 127.007557 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response A 192.168.1.211 > > 424.853464 10.100.0.111 -> 192.168.1.211 SIP/SDP > > Request: INVITE > sip:18005551212@inside.foo.com:5060, > > with session descriptio > > n > > 415.973466 10.100.0.111 -> 192.168.1.211 SIP/SDP > > Request: INVITE > sip:18005551212@inside.foo.com:5060, > > with session descriptio > > n > > > > > ****************************************************************** > > > > How can I get SER to consistently do SRV lookup > after > > rewriting of R-URI? > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

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Re: [Users] Path
by Bogdan-Andrei Iancu 31 Jan '06

31 Jan '06

forgot to cc the list Bogdan-Andrei Iancu wrote: > Hi Andreas, > > you are right - the problem was not the string itself, but the str > itself. Anyhow, hopefully is fixed now. > > regards, > bogdan > > Andreas Granig wrote: > >> Bogdan, >> >>> However, another problem accured. >>> >>> I see in my logs after an INVITE: >>> >>> set_dst_uri: Not enough memory >>> lookup(): Failed to set dst_uri of Path >> >> >> >> In my original patch I made a copy of "route->nameaddr.uri" in >> get_path_dst_uri() of modules/registrar/path.c and then free'd the >> parsed route, but your optimization just points to that uri and also >> free's that route, which invalidates the pointer to that uri, as far >> as I understand. >> >> Can you please check that again, because if I comment out the >> "free_rr(&route)", it does work (although it creates a mem-leak). >> >> Andy > > > >

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[Users] grp reverse function??
by Marcello Lupo 31 Jan '06

31 Jan '06

Hi, there is a way to accomplish the reverse function that group module do? As i know the group module allow me to check if a user is inside a specified group. How can i know in which group is a user without querying the system to know if the user is in all the groups one by one? Actually i'm doing it using avpops and setting an avp that tell me which is the group of the user. I was wandering if there is a more efficient (more performant) way. Thanks, bye, Marcello

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[Users] Is it possiable to rewriting sorc UA name?
by Steve Xu 31 Jan '06

31 Jan '06

Hi All, I found that PSTN GW usually put the "From: xxxx@ipaddress" as ANI (caller ID) when it receive a SIP call. There is a function "rewritingURI" for called URI, I am wandering if there was a function can do rewriting on the the sorc "From" also. For example: user FROM:alice@abc.com make call to hugen(a)abf.com, can I rewrite the alice(a)abc.com to 661234223(a)abc.com when I send INVITE to hugen(a)abf.com ? Any help will be appreciated. Steve

3 3

[Users] privacy=full and contact hf
by Wolfgang Hottgenroth 31 Jan '06

31 Jan '06

Hi, before forwarding a call (INVITE) from the PSTN with flag "privacy=full" in the Remote-Party-ID hf to a none-voice-licensed carrier I need to make sure that the calling party number is completely wiped out from the SIP requests. I see this number at three places: the Remote-Party-ID hf itself, the >From hf and the Contact hf. The Remote-Party-ID hf can simply be dropped or replaced by something anonymous. The From hf can be rewritten using uac_replace_from. But what about the Contact hf? This field is, as far as I can see, required for routing purposes, to tell the called party side where to send replies. So, if I would manage it to rewrite the field, I would have to make sure, the replies sent to the rewritten address once again will be rewritten to target for the meant destination. Is there a best practice for this item? Thanks, Wolfgang

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[Fwd: Re: [Serusers] no SRV query after rewriting R-URI?]
by Klaus Darilion 31 Jan '06

31 Jan '06

I've forgot to cc the list. klaus -------------------------------------------------------------------------- Hi Alan! That's the correct behaviour. Please read RFC 3263. If the URI contains a port, e.g. inside.acme.com:5060 there must not be an SRV lookup. Because the SRV lookup leads to a port, which is already defined. regards klaus On Tue, January 31, 2006 20:28, alan chiu said: > Server: Sip EXpress router (0.8.12 (i86pc/solaris)) > > when endpoint sends 18005551212(a)sip.acme.com, SER > (without rewriting of R-URI, looks it up with SRV > *************** > > 386.656875 192.168.1.246 -> 10.100.0.111 SIP/SDP > Request: INVITE sip:18005551212@sip.acme.com, with > session description > 386.657307 10.100.0.111 -> 192.168.1.246 SIP Status: > 100 trying -- your call is important to us > 88.811658 10.100.0.111 -> 192.168.1.11 DNS Standard > query SRV _sip._udp.sip.acme.com > 98.825208 192.168.1.11 -> 10.100.0.111 DNS Standard > query response, No such name > 98.825496 10.100.0.111 -> 192.168.1.11 DNS Standard > query A sip.acme.com > 108.845136 192.168.1.11 -> 10.100.0.111 DNS Standard > query response, No such name > 108.845468 10.100.0.111 -> 192.168.1.11 DNS Standard > query AAAA sip.acme.com > 118.865215 192.168.1.11 -> 10.100.0.111 DNS Standard > query response, No such name > 416.711408 10.100.0.111 -> 192.168.1.246 SIP Status: > 478 Unresolveable destination (478/TM) > 407.921364 192.168.1.246 -> 10.100.0.111 SIP Request: > ACK sip:18005551212@sip.acme.com > > *********************************************************************************** > > > ************* > after rewritehostport("inside.acme.com:5060"), SER > only does an A lookup > *************** > > 424.851557 192.168.1.246 -> 10.100.0.111 SIP/SDP > Request: INVITE sip:18005551212@nat.foo.com, with > session descripti > on > 415.971953 10.100.0.111 -> 192.168.1.246 SIP Status: > 100 trying -- your call is important to us > 127.006508 10.100.0.111 -> 192.168.1.11 DNS Standard > query A inside.foo.com > 127.007557 192.168.1.11 -> 10.100.0.111 DNS Standard > query response A 192.168.1.211 > 424.853464 10.100.0.111 -> 192.168.1.211 SIP/SDP > Request: INVITE sip:18005551212@inside.foo.com:5060, > with session descriptio > n > 415.973466 10.100.0.111 -> 192.168.1.211 SIP/SDP > Request: INVITE sip:18005551212@inside.foo.com:5060, > with session descriptio > n > > ****************************************************************** > > How can I get SER to consistently do SRV lookup after > rewriting of R-URI? > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] no SRV query after rewriting R-URI?
by alan chiu 31 Jan '06

31 Jan '06

Server: Sip EXpress router (0.8.12 (i86pc/solaris)) when endpoint sends 18005551212(a)sip.acme.com, SER (without rewriting of R-URI, looks it up with SRV *************** 386.656875 192.168.1.246 -> 10.100.0.111 SIP/SDP Request: INVITE sip:18005551212@sip.acme.com, with session description 386.657307 10.100.0.111 -> 192.168.1.246 SIP Status: 100 trying -- your call is important to us 88.811658 10.100.0.111 -> 192.168.1.11 DNS Standard query SRV _sip._udp.sip.acme.com 98.825208 192.168.1.11 -> 10.100.0.111 DNS Standard query response, No such name 98.825496 10.100.0.111 -> 192.168.1.11 DNS Standard query A sip.acme.com 108.845136 192.168.1.11 -> 10.100.0.111 DNS Standard query response, No such name 108.845468 10.100.0.111 -> 192.168.1.11 DNS Standard query AAAA sip.acme.com 118.865215 192.168.1.11 -> 10.100.0.111 DNS Standard query response, No such name 416.711408 10.100.0.111 -> 192.168.1.246 SIP Status: 478 Unresolveable destination (478/TM) 407.921364 192.168.1.246 -> 10.100.0.111 SIP Request: ACK sip:18005551212@sip.acme.com *********************************************************************************** ************* after rewritehostport("inside.acme.com:5060"), SER only does an A lookup *************** 424.851557 192.168.1.246 -> 10.100.0.111 SIP/SDP Request: INVITE sip:18005551212@nat.foo.com, with session descripti on 415.971953 10.100.0.111 -> 192.168.1.246 SIP Status: 100 trying -- your call is important to us 127.006508 10.100.0.111 -> 192.168.1.11 DNS Standard query A inside.foo.com 127.007557 192.168.1.11 -> 10.100.0.111 DNS Standard query response A 192.168.1.211 424.853464 10.100.0.111 -> 192.168.1.211 SIP/SDP Request: INVITE sip:18005551212@inside.foo.com:5060, with session descriptio n 415.973466 10.100.0.111 -> 192.168.1.211 SIP/SDP Request: INVITE sip:18005551212@inside.foo.com:5060, with session descriptio n ****************************************************************** How can I get SER to consistently do SRV lookup after rewriting of R-URI?

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