sr-users February 2006

sr-users@lists.kamailio.org

189 participants
374 discussions

[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] [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] 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] "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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sr-users February 2006