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Authenticating end points by IP
by Ahmed Naji 22 Dec '09

22 Dec '09

People, Please do excuse me if this is a FAQ. I've so far not worked out a way to implement IP authentication effectively. I have a number of gateways/end points/...etc. hitting the switch without registration and originating calls to a number of upstreams I have configured. So far, everything is smooth and FREESwitch is legendary in every way. Over 10,000 simultaneous sessions at 120cps, on a dual-Xeon machine running CentOS and CPU usage is barely a blip. My problem is authentication. I need a method to authenticate end points based only on their IP addresses. I am currently filtering access through the firewalls, but I would really like to delegate this task to FS in prep for an SBC setup I'm working on. If I remove the firewall filters, then anyone is able to get in. I could't workout precisely how ACLs work in FS from the WiKi and the documentation, and I haven't been able to make sense of how Digest functions either. Can anyone shed some light on those two areas ? I would like to really get to the bottom of this and update the WiKi pages with the working setup once done. Regards, Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS

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FreeSWITCH 1.0.5pre10 is now available
by Michael Collins 22 Dec '09

22 Dec '09

It's upgrade-and-test time! The new release announcement is on the main FreeSWITCH page: http://www.freeswitch.org/node/224 Please update, test, and report back bugs and questions. Thanks! -Michael

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How to debug TLS handshake errors?
by Yehavi Bourvine 22 Dec '09

22 Dec '09

Hello, I am trying to debug a TLS handshake error between FreeSwitch and some ATA. When setting the loglevel to 9 I get only a message that TLS handshake failed. Is there some other debug command to show what happens during the TLS handshake process? Thanks! __Yehavi:

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Re: [Freeswitch-users] [Freeswitch-dev] "a1-has" param in gateway setting
by Brian West 22 Dec '09

22 Dec '09

I'm not too sure you can put an a1-hash on outbound auth. /b On Dec 22, 2009, at 11:26 AM, Babak Karvandi wrote: > Hi, > > Does any body know or has test the "a1-hash" parameter with gateway > setting? I am not sure if it is even allowed. I have the following > gateway setting but when the freeswitch starts up it simply ignores this > provider without any error message or attempt to register in the log > file. Thank you for your help in advance. > > <include> > <gateway name="iptel"> > <param name="username" value="MY-USERNAME"/> > <param name="realm" value="iptel.org"/> >  > >  >  > > <param name="a1-hash" value="30f610a85e973f2b29b75ddc1ec3450e"/> > <param name="proxy" value="sip.iptel.org"/> > </gateway> > </include>

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PSTN-to-Internal Call Does Not Get Routed to Voice Mail
by Jerry Richards 22 Dec '09

22 Dec '09

I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry

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Building on Windows
by Dave Stevenson 22 Dec '09

22 Dec '09

Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) 5. How do I specify which options (e.g., mod_flite, to be included iin the build. 6. How do I build the sounds etc. ? Regards Dave 5>os_unix.obj : warning LNK4221: no public symbols found; archive member will be inaccessible 1>filesys.c 6>..\..\pcre\pcre_ucp_searchfuncs.c(158) : warning C4018: '<' : signed/unsigned mismatch 6>..\..\pcre\pcre_ucp_searchfuncs.c(163) : warning C4018: '<=' : signed/unsigned mismatch 16>getenv.c 15>..\..\src\switch_console.c(553) : warning C4244: '=' : conversion from '__w64 int' to 'int', possible loss of data 15>..\..\src\switch_console.c(584) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data 15>switch_core_media_bug.c 15>..\..\src\switch_core_media_bug.c(178) : warning C4244: '=' : conversion from 'switch_size_t' to 'uint32_t', possible loss of data 15>..\..\src\switch_core_media_bug.c(221) : warning C4244: '=' : conversion from 'switch_size_t' to 'uint32_t', possible loss of data 15>..\..\src\switch_core_media_bug.c(222) : warning C4244: '=' : conversion from 'switch_size_t' to 'uint32_t', possible loss of data 18>nta_tag.c 21>c:\freeswitch src\freeswitch\libs\xmlrpc-c\src\xmlrpc_server_abyss.c(894) : warning C4047: 'initializing' : 'TOsSocket' differs in levels of indirection from 'void *' 31>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler 86>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler 86>mod_opal.cpp 86>c:\freeswitch src\freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error C1083: Cannot open include file: 'ptlib.h': No such file or directory

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Setting Restrictions on Default Dialplan
by Edmar Cruz 21 Dec '09

21 Dec '09

Hi Sir, How can I allow international calling in the dialing plan but for select accounts only? For example i want to restrict 8555555 to call this ip address 182.138.252.12 using the default configuration.. Does this command should be put in the default.xml or in the default folder and the filename is 00_restict.xml? <extension name="dialsample"> <condition field="destination_number" expression="^(8555555)$"\> <action application="set" data="effective_caller_name=${effective_caller_id_name}"/> <action application="set" data="effective_caller_number=${effective_caller_id_number}"/> <action application="set" data="hang_up_after_bridge=true"/> <action application="bridge" data="sofia/default/$1(a)182.138.252.12"/> </condition> </extension> When i tried this command both of them nothing happen 8555555 can call 182.138.252.12 i want it to restrict this account for not calling 182.138.252.12.. Please help.. Thanks, Edmar -- View this message in context: http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26868725p2… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Skypiax: Skype account frozen
by 大泥人 21 Dec '09

21 Dec '09

Hello, I noticed some guys had develop the Skype module while there is a policy from Skype(Ulimited call planso call FAP: ):"Each subscription is to be used by one person only and is not to be shared with any other user (whether via a PBX, call centre, computer or any other means)" , which means once you use Skype unlimited calls plan into PBX, Skype will frozen your account without any money return. That's a big risk for anybody to use Skype unlimited call plan. My question is how do we avoid such kind of risk? Thanks Daniel Zeng From: freeswitch-users-request(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 42, Issue 193 To: freeswitch-users(a)lists.freeswitch.org Date: Mon, 21 Dec 2009 08:21:08 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --附转发的邮件-- From: anthony.minessale(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Mon, 21 Dec 2009 09:42:30 -0600 Subject: Re: [Freeswitch-users] Difference between ESL execute() andexecuteAsync() if you run the socket in async mode, every call to execute is async if you don't specify async in the socket app in FS all calls are synchronous but you can send async calls with te asyncExecute On Sat, Dec 19, 2009 at 9:16 PM, Ron McLeod <ron.freeswitch(a)mcleodnet.com> wrote: Here's the ES network trace: Content-Length: 1502 Content-Type: text/event-plain Event-Name: CHANNEL_STATE Core-UUID: bb9ea62a-ed02-11de-91b1-8b7cb185f66f FreeSWITCH-Hostname: ron-laptop FreeSWITCH-IPv4: 192.168.100.132 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-12-19%2019%3A12%3A09 Event-Date-GMT: Sun,%2020%20Dec%202009%2003%3A12%3A09%20GMT Event-Date-Timestamp: 1261278729767397 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_perform_set_running_state Event-Calling-Line-Number: 1024 Channel-State: CS_ROUTING Channel-State-Number: 2 Channel-Name: sofia/internal/699%40192.168.100.132 Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: ringing Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 699 Caller-Dialplan: XML Caller-Caller-ID-Name: Ron%20Soft%20Phone Caller-Caller-ID-Number: 699 Caller-Network-Addr: 192.168.100.3 Caller-Destination-Number: 444 Caller-Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f Caller-Source: mod_sofia Caller-Context: mytest Caller-Channel-Name: sofia/internal/699%40192.168.100.132 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1261278729764077 Caller-Channel-Created-Time: 1261278729764077 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f call-command: execute execute-app-name: answer execute-app-arg: Content-Type: command/reply Reply-Text: +OK sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f call-command: execute execute-app-name: playback execute-app-arg: /tmp/ann.wav Content-Type: command/reply Reply-Text: +OK > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch- > users-bounces(a)lists.freeswitch.org] On Behalf Of Ron McLeod > Sent: Saturday, December 19, 2009 5:30 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] Difference between ESL execute() > andexecuteAsync() > > I don't notice any different in behavior between execute() and > executeAsync(). I was expecting that executeAsync() would return > right-away, and that execute() would only return after the specified > application runs to completion (CHANNEL_EXECUTE_COMPLETE event). > > Running the sample app below, I see the "About to call execute(playback)" > and "returned" displayed one right-after the other, even though the file > being played takes about 4 minutes to play-out. > > Do I have this wrong, or is there something incorrect in my app? > > APP: > #!/usr/bin/php > <?php > require_once "ESL.php"; > > $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon'); > $eventSocket->events('plain', 'CHANNEL_STATE'); > $eventSocket->filter('channel-state', 'CS_ROUTING'); > > // Wait for new call attempts > while($eventSocket->connected()){ > $event = $eventSocket->recvEvent(); > $serializedBody = $event->serialize(); > $listOfLines = toArrayOfLines($serializedBody); > $nameValuePairs = toArrayOfNameValuePairs($listOfLines); > > $uuid = $nameValuePairs['Caller-Unique-ID']; > printf("New call from uuid: $uuid\n"); > > // answer the caller and play announcement > $eventSocket->execute('answer', Null ,$uuid); > > printf("About to call execute(playback)\n"); > $eventSocket->execute('playback', '/tmp/ann.wav', $uuid); > printf("returned\n"); > } > ?> > > > DIALPLAN: > <?xml version="1.0" encoding="utf-8"?> > <include> > <context name="mytest"> > <extension name="mytest"> > <condition field="destination_number" expression="^.*$"> > <action application="park"/> > </condition> > </extension> > </context> > </include> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > This email was Anti Virus checked by Astaro Security Gateway. > http://www.astaro.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 --附转发的邮件-- From: andrew(a)hijacked.us To: freeswitch-users(a)lists.freeswitch.org Date: Mon, 21 Dec 2009 11:07:08 -0500 Subject: Re: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: > Hi > > its good to hear > > any compare document between Vicidial and this project > No document, but briefly: * More focused on inbound than on outbound (at least for the moment) vicidial is more geared for outbound. * Handles email in queue (and soon chat), vicidial is only voice. * wrapup time is per-call not static per-'campaign' * license is a little more liberal * can operate as a distributed system * doesn't need asterisk ;) Andrew --附转发的邮件-- From: jbr(a)consiglia.dk To: freeswitch-users(a)lists.freeswitch.org Date: Mon, 21 Dec 2009 17:13:21 +0100 Subject: [Freeswitch-users] mod_xml_curl and gateways I wonder if it is possible to define common gateways (not user specific gateways) by xml_curl, and if so, the bindings and syntax to use? All the best /Jon --附转发的邮件-- From: mrene_lists(a)avgs.ca To: freeswitch-users(a)lists.freeswitch.org Date: Mon, 21 Dec 2009 11:21:04 -0500 Subject: Re: [Freeswitch-users] mod_xml_curl and gateways Hi, All gateways are common, putting them in a user only serves the purpose of grouping related information together in the XML files. This said, you can bind to the "configuration" section and return those gateways as part of the sip profile's xml data. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene(a)avgs.ca On 21-Dec-09, at 11:13 AM, Jon Bruel wrote: I wonder if it is possible to define common gateways (not user specific gateways) by xml_curl, and if so, the bindings and syntax to use? All the best /Jon _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________ 上Windows Live 中国首页，下载Messenger2009安全版！ http://www.windowslive.cn

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ACLs through proxy
by Bill W 21 Dec '09

21 Dec '09

Hi All, I have a FreeSWITCH cluster behind an OpenSIPS proxy/load balancer, and I'd like to be able to use the auth-calls feature in my sip profile in conjunction with the <param name="auth-acl" value="1.2.3.0/8"/> parameter in the directory. In addition to running the INVITEs through the load balancer, I also need to run the REGISTERs through the load balancer because some of my endpoints are behind NAT firewalls, and therefore won't accept incoming calls from IPs other than the IP they registered to. INVITEs from the cluster going to registered endpoints are sent back through the proxy, thereby solving the NAT problem. However, having the proxy in the path effectively negates using IP based ACLS. The functionality I require is as follows: 1. Only allow registration if the endpoint IP matches it's own unique acl CIDR (specified in the directory). 2. Only accept INVITEs from endpoints that authenticate AND match the acl CIDR (again, specified in the directory). Does anyone have any recommendations on the best way to get the auth-calls functionality using an IP other than the IP of the last hop? If not, how hard would it be to add a feature to the auth-calls parameter to accept a channel variable from which to obtain the actual endpoint IP? Thanks! Bill

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mod_xml_curl and gateways
by Jon Bruel 21 Dec '09

21 Dec '09

I wonder if it is possible to define common gateways (not user specific gateways) by xml_curl, and if so, the bindings and syntax to use? All the best /Jon

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