- FreeSWITCH-users - FreeSWITCH Mailing Lists

how to ban this spammer?
by Neo Haux 04 Jun '14

04 Jun '14

Hi all, I am receiving hundreds of INVITE/minute and in the log I can see: /2014-06-04 13:52:30.189371 [WARNING] sofia_reg.c:1532 SIP auth challenge (REGISTER) on sofia profile 'internal' for [340@MyExternalIP] from ip 62.210.142.39// //2014-06-04 13:52:42.789530 [WARNING] sofia_reg.c:1532 SIP auth challenge (REGISTER) on sofia profile 'internal' for [341@MyExternalIP] from ip 62.210.142.39// //2014-06-04 13:52:55.479999 [WARNING] sofia_reg.c:1532 SIP auth challenge (REGISTER) on sofia profile 'internal' for [341@MyExternalIP] from ip 62.210.142.39// //2014-06-04 13:53:08.289660 [WARNING] sofia_reg.c:1532 SIP auth challenge (REGISTER) on sofia profile 'internal' for [342@MyExternalIP] from ip 62.210.142.39// //2014-06-04 13:53:21.679512 [WARNING] sofia_reg.c:1532 SIP auth challenge (REGISTER) on sofia profile 'internal' for [342@MyExternalIP] from ip 62.210.142.39/ In the /etc/fail2ban/filter.d/freeswitch.conf file I have these lines: /failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure $REGISTER$ on sofia profile \'\w+\' for \[.*\] from ip <HOST> \[WARNING\] sofia_reg.c:\d+ SIP auth failure $INVITE$ on sofia profile \'\w+\' for \[.*\] from ip <HOST>/ You can see clearly that my logs contain failure word not "auth challange". My question is : If I put "auth challange" in my /etc/fail2ban/filter.d/freeswitch.conf will I block regular known and authenticated SIP clients ? If yes, could you help find the right regex to stop this kind of spammers ? Thank you very much in advance.

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query database for VM active greeting number?
by Peter Hartmann 04 Jun '14

04 Jun '14

Hello, I'm trying to find a users active greeting number. See the tl:dr below for big picture. I thought maybe vm_fsdb_pref_greeting_get would be what I'm looking for but I think that may only return the filename of a specific greeting where [slot] is the greeting number. Although I wasn't able to get this command to work for some reason...."profile not found" using 'default'. Could anyone tell me if I can do this within FS or do I need to run a query externally on the DB. By the way I'm using the default sqlite db. tl:dr Ultimately I'm looking to leverage the existing VM greeting recording functionality to enable an end user to add important announcements before the regular IVR begins. Once the active greeting is in a variable, the IVR would reference a wav [absolute-path]/greeting_${active_greeting_number}.wav. When not in use, greeting 9 would be active, referencing a non-changing wav with a ~a millisecond of silence so as to not break the IVR. Thanks much! Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com (212)203-8870 If I can't explain it to you in plain language, that means I don't understand it.

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Come to ClueCon early.... The OpenSIPS Summit is back!
by Alex Goulis 04 Jun '14

04 Jun '14

*It's officially less than two months before the OpenSIPS summit hits Chicago!* If you've ever wondered how OpenSIPS can help you with your Freeswitch deployments, now's your chance. For all of you attending ClueCon, we'll be holding an OpenSIPS summit on Aug 4 running most of the day before ClueCon officially kicks off. Last year's summit was a tremendous success. Everyone that attended enjoyed interesting topics and discussions. Along with this announcement, we would like to announce the first two speakers for this years summit: *Using OpenSIPS as a single entry point for a SIP network * /Nowadays, it is becoming harder to protect VoIP providers. The complexity of the environment makes it easy to miss a security breach. Patching all equipment is usually not viable with the current workmanship. What OpenSIPS can do to help in this battle? In this presentation, we are going to cover: Using OpenSIPS as a single point of protection. / *Flavio E. Goncalves* is a CTO of SipPulse, a company dedicated to implement solutions for VoIP providers and High Volume Telephony. He is the author of the book Building Telephony Systems with OpenSIPS. *Advanced SIP Routing using OpenSIPS *** /The goal of this presentation will be to provide you enough information to be able to make an informed business decision regarding which advanced SIP routing module (or modules) to use. Business rules for outbound calls will likely be different than for inbound calls. An understanding of the modules used for advanced SIP routing will allow you to make the best decision based on your particular use case./ *Norman Brandinger* is an expert in the configuration and use of OpenSIPS, FreeSWITCH and related VoIP technologies. He has successfully deployed VoIP solutions in the Mobile App, Hospitality and Financial Services sectors resulting in new business opportunities and reduced operating expenses. Norm is currently the VP of Engineering for textPlus. You can get more information about the summit at: http://www.opensips.org/Community/Summit-2014Chicago Registration is still open: http://www.opensips.org/Community/Summit-Registration Last year's summit highlights: http://www.opensips.org/Community/Summit-2013Chicago

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ClueCon Weekly! Markus Lindenberg of GOfax.IP
by Ken Rice 04 Jun '14

04 Jun '14

Hey Y¹all! Join us today at 1PM EDT, 10AM PDT For ClueCon Weekly! This weeks guest is Markus Lindenberg of GOFax.IP! A new faxing Platform built around FreeSWITCH! Call sip:888@conference.freeswitch.org or +1-919-386-9900 or for more access info see, https://confluence.freeswitch.org/display/FREESWITCH/Community Have Fun and hear you there! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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PHP ESL.so not useable with PHP 5.5 on FreeBSD
by Daniel Neubert 04 Jun '14

04 Jun '14

Hello everybody, I already opened a JIRA (https://jira.freeswitch.org/browse/ESL-97) for this issue but got no reply so far and I wonder if anybody else has this problem: We tried to build the ESL module for PHP on FreeBSD 9.1 and 10.0 but we always end up in sth. like that: /usr/local/lib/php/20121212/ESL.so: Undefined symbol "_ZN13ESLconnectionC1EPKcS1_S1_" We've tested gcc and clang (FreeBSD 9.2: gcc 4.2.1, FreeBSD 10.0: clang 3.4.1, gcc 4.2.1. Tested gcc4.7 with binutils 2.24.) but without any success. Any suggestions? Greetings from Dortmund, Germany Daniel

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User Not found
by Iwada Eja 04 Jun '14

04 Jun '14

Hi, Im Completely new to Freeswitch and appreciating it Im trying to dynamically register a user using mod_lua, This is the returned XML from my lua script <include> <user id="9090"> <params> <param name="password" value="9090"/> <param name="vm-password" value="7000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="7000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Jude"/> <variable name="effective_caller_id_number" value="Jude"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> however each time i try to make a call with that user, i get a " Can't find user [9090@My_IP] from My_IP You must define a domain called 'My_ip' in your directory and add a user with the id="9090" attribute and you must configure your device to use the proper domain in it's authentication credentials. " However, if i manually create a 9090.xml file adding the same contents, everything works as should this is my lua.conf <param name="xml-handler-script" value="dp.lua"/> <param name="xml-handler-bindings" value="directory"/> Any help would is highly appreciated!. Thank You --

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SpiderMonkey: undefined symbol: switch_core_hash_insert
by Mark Campbell-Smith 04 Jun '14

04 Jun '14

Hi, I'm trying to get SpiderMonkey to work, but I'm getting the error: 2014-06-03 22:50:46.347438 [CRIT] switch_loadable_module.c:1448 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/mod/mod_spidermonkey.so: undefined symbol: switch_core_hash_insert** I'm running FreeSWITCH Version 1.5.13b+git~20140603T002504Z~6722e3dfea~64bit (git 6722e3d 2014-06-03 00:25:04Z 64bit)

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Re: [Freeswitch-users] kamailio+freeswitch+HA (?????)
by Luis Azedo 04 Jun '14

04 Jun '14

> > try kazoo. https://github.com/2600hz/kazoo > > ---------- Forwarded message ---------- > From: Денис <denis(a)ringme.ru> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Tue, 03 Jun 2014 19:56:02 +0400 > Subject: [Freeswitch-users] kamailio+freeswitch+HA > Hello. > > We try to make fault-tolerant system, but not all modules are able to > request configs through mod_xml_curl eg callcenter can not receive queues > and agents, and maintain the status queue. We want to do freeswitch without > a local configuration. > And what better to use the base, we still think couchdb or Cassandra > > Who can share experiences or advice? > > > > >

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Re: [Freeswitch-users] G.729 Passthrough question
by Guillermo Ruiz Camauer 04 Jun '14

04 Jun '14

Steve, Yes, all surveys are handled by DTMF with the addition of some open ended questions (not more than one or two per survey). I only use RFC2833 for DTMF, since my SIP providers require that I use G.729. The open ended questions just get recorded and someone listens to them and codes them later. DTMFs can be a pain, since some upstream providers might not support them well. Because of the call volume I have, many times. I have alerted my Sip providers as to misconfigured routes they have with their peers. My system has 2670 call channels, most on E1 ISDN with Dialogic cards, but 900 of those are handled by FS, since I have stopped buying E1/Dialogic 3-4 years ago and now only grow my system with VoIP and FS. I would be glad to answer any questions regarding DTMF or anything else related to this system. Regards, Guillermo Sent from my iPhone > On 03/06/2014, at 22:49, Steve Underwood <steveu(a)coppice.org> wrote: > > Hi Guillermo, > > Are your 900 calls handling surveys by DTMF? If so I would be interested > to hear your experiences with the reliability of that. We still see so > many quirks with the handling of DTMF in VoIP applications that > experiences with such a large deployment are of value. Especially ones > using low bit rate codecs like G.729, where audio DTMF can be quite > troublesome. > > Regards, > Steve > > >> On 06/04/2014 04:06 AM, Guillermo Ruiz Camauer wrote: >> Brian, >> >> Thanks for the reply. Then if I understand correctly, you only need a >> G.729 license if the audio stream will get transcoded to something >> else? I realize that the licesnces are only $10 each, but I also have >> another system which does polling (surveys) which handles 900 >> simultaneous calls. >> >> Thanks, >> >> Guillermo >> >> >> On Tue, Jun 3, 2014 at 11:01 AM, Brian West <brian(a)freeswitch.org >> <mailto:brian@freeswitch.org>> wrote: >> >> You could, but licenses are cheap enough that you can buy a hand >> full for this purpose. >> -- >> Brian West >> brian(a)freeswitch.org <mailto:brian@freeswitch.org> >> >> >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Jun 2, 2014, at 5:38 PM, Guillermo Ruiz Camauer >> <grcamauer(a)gmail.com <mailto:grcamauer@gmail.com>> wrote: >> >>> If I set up a FS server as a standalone VoiceMail server, record >>> all voicemail prompts in G.729, and record all user voice >>> messages in G.729, can the server be set up in G.729 passthrough, >>> or must it have full G.729 licenses? The point being that there >>> will never be any transcoding to other codecs. >>> >>> Thanks in advance, >>> >>> -- >>> Guillermo >>> ____________ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Guillermo Ruiz Camauer >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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

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***SPECIAL ANNOUNCEMENT*** The FreeSWITCH Team has Deployed Crowd!
by Ken Rice 04 Jun '14

04 Jun '14

Greetings FreeSWITCHers! The FreeSWITCH Team is happy to announce the deployment of Crowd! Crowd is a central user database that allows us to now support that same logins across most of the FreeSWITCH.org infrastructure that is used to support the project. Currently this includes Jira, Fisheye, and Confluence. In the near future, this will include other things like www.freeswitch.org, Stash (which is similar to Github) and other things we have in the works. What this means for you today, if you have been using any of the Atlassian software products on the FreeSWITCH.org domain your jira passwords should now work. (Don¹t be surprised if you are don¹t have access to something, we¹re still working on that part). However if you can not login to Jira or other areas you have previously been able to login to, please clear any cookies in your browser for the freeswitch.org domain and in particular any cookies for ³crowd². (In Firefox go to the preferences/settings screen then to the privacy tab, click the link to ³remove individual cookies², in the searchbox enter freeswitch and delete the results related to freeswitch.org, then enter crowd and delete those) If clearing the cookies (and possibly the cache) does not work for you, please contact me offlist or on irc (SwK in #freeswitch on irc.freenode.net) See you at ClueCon! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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