June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

set ptime and handle "a=sendonly"
by Dmitriy Shumaev 05 Jun '14

05 Jun '14

Hi. 1. I'm trying to send 40 ms of speech in each RTP frame when working with a specific/ certain gateway . By means of modifying [var.xml]: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA@8000h@40i,PCMU@8000h@40i"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA@8000h@40i,PCMU@8000h@40i"/> , it is possible to call FROM this gateway (gateway does not send ptime in SDP). But, when the call comes from the other client with ptime set to 20 in SDP, outgoing INVITE messange TO this gateway contains ptime set to 20. How can I set ptime to 40 ( it's desirable to do so only for one specific gateway ). 2. I'm using FS for PTT (Push-To_Talk) services. 2.1 When FS gets incoming INVITE / reINVITE with "a=sendonly" in SDP and responde with "a=recvonly" it MUST *NOT* send RTP packets to UAC, but FS does so. So, how can I fix this? 2.2 To start talking FS sends reINVITE (I use "uuid_media_reneg __uuid" command for this). I need to add "a=sendonly" to this reINVITE. How it can be done? With best regards, Shumaev D.A., programmer @ KBR Ltd.

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Hardware for Freeswitch setup
by Oscar Franzen 05 Jun '14

05 Jun '14

Hi! I'm looking for some help with hardware for a production setup of freeswitch. I need at least 150 concurrent calls, but would prefer to buy a server which can handle more to facilitate growth. I am looking at using the same codec (i think it is g711) for both legs of the call. I want to record all calls. Either to whatever codec and then transcode them separately or directly to mp3/ogg. I am going to use ESL to control FSW and also listen to most events over the socket. What would you recommend for hardware? Is it recommended to buy a sound card for transcoding or just buy enough CPU? I will probably run from SSD, record to same SSD or other local disk and continuously move recordings to external (network) storage. So: 150 concurrent calls ~10 initiations/second mod_event_socket - for control and events A limited amount of eavesdropping recording (either to original codec or ogg/mp3) Some "standard" modules. Regards Oscar

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[ERR] switch_rtp.c:839 Invalid STUN/ICE packet received
by rockxsj 05 Jun '14

05 Jun '14

Hi,everyone.When I look at the log of freeswitch,I can see many logs like this: fs.log.2014-06-05-00-00-01.1:ce9e4f0c-ebfc-11e3-ad0d-955082dbd7ee 2014-06-04 23:28:38.337404 [ERR] switch_rtp.c:839 Invalid STUN/ICE packet received 112 bytes My freeswitch is not behind NAT,and my client is chrome browser.Can I modify some config to disable stun/ice or other method to solve this problem?

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DTMF Issue
by Richard Mace 05 Jun '14

05 Jun '14

Hi All, I have a FreeSWITCH server running behind a firewall and can successfully connect a Jitsi software client to the server remotely. I can dial 5000 from Jitsi and hear audio, but when I dial a DTMF digit,nothing happens. I have tried configuring Jitsi to use different DTMF methods, but none seem to work. It is a vanilla install of FreeSWITCH, Running on Debian 7 installed via the packages so the config files will be default. Any ideas? Thanks Richard

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Postgres support in Core
by Iwada Eja 04 Jun '14

04 Jun '14

Please how do i use Postgre in Freeswitch Core. I've been reading the wonderful Freeswitch 1.2 Book. Examples there on connecting to a database. It uses LauScript for the database example - loading luasql.postgres. It's my thought that since Postgres is now in the core, the example might not be a best practice again. I ./configure --enable-core-pgsql-support to enable Postgres Support and edited my switch.conf.xml. I've also seen a couple of freeswitch.Dbh examples but non with Postgre Core. Please can someone point me to the right Docs or offer some directions. Thank You --

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Re: [Freeswitch-users] how to ban this spammer?
by Neo Haux 04 Jun '14

04 Jun '14

Thanks Michael, May be it's related to SIP not freeswitch, but how can a sip client ask a FS to make an external call without authentication. Shouldn't be banned from making any request by FS if it doesn't find this user already registred ? Here is my list_users: freeswitch@internal> list_users userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number 100|default|192.168.1.1|default|sofia/internal/sip:100@192.168.1.1:5060||| 101|default|192.168.1.1|default|sofia/internal/sip:101@192.168.1.1:5061||| 102|default|192.168.1.1|default|error/user_not_registered||| 103|default|192.168.1.1|default|error/user_not_registered||| 104|default|192.168.1.1|default|error/user_not_registered||| Subject: Re: [Freeswitch-users] how to ban this spammer? From: Michael Jerris <mike(a)jerris.com> Date: 14-06-04 02:20 PM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> yes, if you blocked everything that was challenged you would probably block legitimate traffic. There is no "regex" that can tell you the difference between good and bad traffic like this, perhaps something that looks more specifically at traffic patterns could help, but that would be significant logic to find the right mix. you could do something with iptables for rate limiting that can minimize the effectiveness of attacks like this. Mike On Jun 4, 2014, at 5:59 PM, Neo Haux <neo.haux(a)gmx.com <mailto:neo.haux@gmx.com>> wrote: 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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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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