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Is this SMS scenario possible in FreeSWITCH?
by Mario G 05 Jun '14

05 Jun '14

I checked the wiki and ML for this but could not determine if this is possible: Someone sends an SMS to an ITSP number into FreeSWITCH, then have FreeSWITCH forward the SMS to an outside number. I want to do with SMS what I now do with a bridge to cell phones. My current ITSP does not yet support SMS but would lie to make this work someday if possible. Thanks. Mario G

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FreeSWITCH 1.4.6 Released!
by Ken Rice 05 Jun '14

05 Jun '14

FreeSWITCH 1.4.6 has been released to address a few bugs. This bugs include a regression in re-invite parsing, a few more issues uncovered by our continuing Coverity scans of the code base, addressing some build issues mod_perl, and addressing a WebRTC issue with Chrome where it now requires a longer DTLS Password. This is a routine maintenance update release. Source Tarball is available at http://files.freeswitch.org/freeswitch-1.4.6.tar.gz Additional information on obtaining and installing FreeSWITCH is located at https://confluence.freeswitch.org/display/FREESWITCH/Installation -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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freeswitch how does mod_callcenter deal with 302 redirect
by Andrew Cassidy 05 Jun '14

05 Jun '14

As subject suggests, how does mod_callcenter deal with a 302 redirect from an agent's phone? We have a number of staff out of the office and they have set up call forwarding rather than logging out of the queue, just wondering if it's likely to cause issues. -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F <info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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OpenSSL vulnerabilities.
by Michael Jerris 05 Jun '14

05 Jun '14

> A whole list of new vulnerabilities were announced today: > > https://www.openssl.org/news/secadv_20140605.txt > > We recommend everyone using openssl for anything upgrade to 1.0.1h immediately. > >

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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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