April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SSL/TLS customized encryption.
by mehroz 28 May '13

28 May '13

Hi all, I am trying to create certificates and keys on a customized encryption scheme. AFAIK gentls_cert generates certificates and keys with : Public Key Info: Public Key Algorithm: rsaEncryption Public-Key: (2048 bit) I want the encryption to be according the SUITE-B standard ,i .e 1. (AES) with key sizes of 128 and 256 bits 2 . Elliptic Curve Digital Signature Algorithm 3 . Elliptic Curve Diffie–Hellman (ECDH) — key agreement 4 .Secure Hash Algorithm 2 (SHA-256 and SHA-384) — message digest I can see FS script (gentls_cert) is using openssl at the back. And as i have very little idea of encryption and all using openssl , i need to know if this is possible to mend this script to our requirements, if so ... how and where should i look at first! Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-TLS-customized-encryption… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 6

inband DTMF bleeding through
by Matt Broad 23 May '13

23 May '13

Hi, I have the following dialplan in my public folder (the dest_number below is just an example): <include> <extension name="dialExt1002"> <condition field="destination_number" expression="^(1234)$"> <action application="set" data="drop_dtmf=true"/> <action application="export" data="drop_dtmf=true"/>  <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ignore_early_media=true" /> <action application="set" data="ringback=${uk-ring}"/> <action application="set" data="call_timeout=30"/> <action application="bridge" data="user/1002"/> </condition> </extension> </include> I am trying to prevent the DTMF tones from being heard by the b-leg (in this case user/1002). This works fine if the extension is called by a registered softphone (the user_context is set to public). The digits are pressed and no beeps or tones are heard by the other side. If this extension is dialed by a landline/mobile number, when a digit is pressed there is a slight "bleeding" of the DTMF. Enough to distinguish between different digits, though probably not enough to determin the digit pressed. I have tried several things such as stop_dtmf in the dialplan, setting the dtmf-duration to 0 (not quite sure what this param does but tried it anyway :) ) in the sip profile but nothing seems to prevent it. Could it be our provider? They have confirmed that they use out-of -band rfc2883. But without the drop_dtmf being set in the dialplan the full tone is heard. Could they be sending inband and out-of-band? Could anyone suggest any tests I could run or any log files I could look at please? Many Thanks Matt

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Newbie's installation: Can't start FS at boot
by Dyna Guy 23 May '13

23 May '13

I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh install) and followed the instruction on the wiki installed the new Freeswitch v1.0.3. After the installation I can start FS by issus command "/usr/local/freeswitch/bin/freeswitch". After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and chmod it to 755. Reboot the server and FS is not running. If I try the start up script: [root@localhost build]# /etc/init.d/freeswitch start Starting freeswitch: [ OK ] [root@localhost build]# /etc/init.d/freeswitch status freeswitch dead but subsys locked Did I miss somthing here? Please help. Thanks a lot. dynaguy Here is the copy of the origenal freeswitch.init.centos5 ---------------------------- [root@localhost ~]# cat /etc/init.d/freeswitch #!/bin/bash # # /etc/rc.d/init.d/freeswitch # # The FreeSwitch Open Source Voice Platform # # chkconfig: 345 89 14 # description: Starts and stops the freeswitch server daemon # processname: freeswitch # config: /usr/local/freeswitch/conf/freeswitch.conf # pidfile: /usr/local/freeswitch/log/freeswitch.pid # # Source function library. . /etc/init.d/functions PROG_NAME=freeswitch PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_USER=${FS_USER-freeswitch} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} LOCK_FILE=/var/lock/subsys/freeswitch FREESWITCH_ARGS="-nc" RETVAL=0 # Source options file if [ -f /etc/sysconfig/freeswitch ]; then . /etc/sysconfig/freeswitch fi # <define any local shell functions used by the code that follows> start() { echo -n "Starting $PROG_NAME: " if [ -e $LOCK_FILE ]; then if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then echo echo -n $"$PROG_NAME is already running."; failure $"$PROG_NAME is already running."; echo return 1 fi fi cd $FS_HOME daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1" echo RETVAL=$? [ $RETVAL -eq 0 ] && touch $LOCK_FILE; echo return $RETVAL } stop() { echo -n "Shutting down $PROG_NAME: " if [ ! -e $LOCK_FILE ]; then echo echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running." failure $"cannot stop $PROG_NAME: $PROG_NAME is not running." echo return 1; fi cd $FS_HOME $FS_FILE -stop > /dev/null 2>&1 killproc $PROG_NAME RETVAL=$? echo [ $RETVAL -eq 0 ] && rm -f $LOCK_FILE; return $RETVAL } rhstatus() { status $PROG_NAME; } case "$1" in start) start ;; stop) stop ;; status) status $PROG_NAME RETVAL=$? ;; restart) stop start ;; reload) # <cause the service configuration to be reread, either with # kill -HUP or by restarting the daemons, in a manner similar # to restart above> ;; condrestart) [ -f $PID_FILE ] && restart || : ;; *) echo "Usage: $PROG_NAME {start|stop|status|reload|restart}" exit 1 ;; esac exit $RETVAL

3 2

advise to troubleshoot a call
by Philippe Le Toquin 18 May '13

18 May '13

Hello, Recently I (well my wife) is having voice problem when she is on a call. Typically she can hear people but they can't The problem is that it is not all the time and when it happens it last a few seconds then all back to normal... then later on it happens again. I am still in the process of trying to find if there is a pattern in the problem but I would like to be able to look at the log from FS. What would you recommend to be able to look at the log (what debug level). Any tips to make it easier to identify an issue in the log (like message to look for?) Thanks Philippe

5 13

Limit concurrent calls
by Tomasz Szuster 14 May '13

14 May '13

Hello, I've set up limit for outgoing calls using http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01, especially by adding: limit hash fraud_protection calls_max_intl 32 !NORMAL_TEMPORARY_FAILURE to the dialplan. When I created second dialplan with different call_max_intl value freeswitch did not get it. It has still remember call_max_intl 32 from first dialplan. At console I've got: *Usage for fraud_protection_calls_max_intl is already at max value (32)* I've also try with: <action application="limit_execute" data="hash outbound gateway 12 bridge sofia/gateway/gateway/$1" /> Result was the same as for limit hash. My goal is to create dialplans with different calls max where calls max value are treated separately. Can you please advice how to proceed, where to look ? -- Regards. Tom

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Re: [Freeswitch-users] Does freeswitch support SIP-T/SIP-T?
by xiaofengcanyuexp＠163.com 11 May '13

11 May '13

Dear freeswitch support I have been studying freeswitch for a few weeks. I will be planning to take the freeswitch as a signal gateway connecting PSTN(ISUP) and SIP. It needs convert the ISUP to SIP based on RFC3372/RFC3204/RFC3398 and vice versa. I notice in the mime_type.cfg supporting applicaiton/ISUP, but I don't find any code in sofia(SIP) module to decode/encode the application/ISUP. My question is: Does freeswitch support to do the signalling gateway connecting ISUP(freeTDM module also has the MTP layer support) and SIP(SIP-T/SIP-I)? If yes, could you let me know how it works? Thank you so much! Windy ------------------- 2013-03-24 = = = = = = = = = = = = = = = = = = = =

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Re: [Freeswitch-users] Freeswitch silent crash
by Mr Nathan Downes 09 May '13

09 May '13

I appear to have fixed this on mine, when I moved everything over to the new server (12.0.4) I forgot the ulimit stuff in the init.d script and threading=0 in the odbc file for postgres.. Since these have gone in I haven't had a crash.. I also moved up to current head from stable.. not sure if this helps. From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Zheng Sent: Monday, 29 April 2013 4:08 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch silent crash I have this problem too. There are several similarity here. There is no core dump even with suid_dumpable to 1 and other suggestions on the wiki, e.g. ulimit, change directory to /tmp in FS start script. Using ubuntu latest 12.04, running as non-root, lua to generate dialplan and control routing. The most troubling/headache is that there is no core dump when crash. I understand that there is no much to do without it... Any suggestion is appreciated! On Fri, Apr 26, 2013 at 3:44 AM, Antoine Reversat <antoine(a)fibrenoire.ca> wrote: Sure thing, hopefully it'll crash again soon. On Fri, Apr 26, 2013 at 9:43 AM, Brian West <brian(a)freeswitch.org> wrote: Please attach the backtrace of the core file and not the core file itself. On Apr 26, 2013, at 7:50 AM, Antoine Reversat <antoine(a)fibrenoire.ca> wrote: > I haven't created a Jira yet, I'm waiting to have a core dump to attach to it. If you create a one please let me know. > -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 <tel:%2B1.918.420.9266> | F: +1.918.420.9267 <tel:%2B1.918.420.9267> | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 <tel:%2B883%205100%201420%209266> ISN: 410*543 _________________________________________________________________________ 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 -- Antoine Reversat Fibrenoire - <http://www.fibrenoire.ca/> www.fibrenoire.ca A: 550 ave Beaumont, suite 320, Montreal, Qc, H3N 1V1 T: 514 907-3002 x132 <tel:514%20907-3002%20x132> antoine <mailto:jf@fibrenoire.ca> @fibrenoire.ca _________________________________________________________________________ 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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Dead channels in freeswitch
by Farrukh Ali 09 May '13

09 May '13

Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply

9 33

External Softphone vs. Internal Question
by Jeff Bernhardt 09 May '13

09 May '13

Hi. I have the following basic setup questions: When using a softphone (Bria on iPhone) from external (on a different external ip address), I could register but no audio would be passed either way for any calls. I saw that I should set ext-rtp-ip in the internal sip profile to my external ip address (it was on auto-nat, which apparently wasn't working) in this wiki http://wiki.freeswitch.org/wiki/NAT_Traversal That didn't work, so I also set my ext-sip-ip to my public ip. After that, I could pass audio. However, if I register the phone internally instead and call for instance the IVR test line, the call drops after 30 seconds. So it's either no audio when registered externally or 30 second calls when registered internally. I found this wiki: http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios I fall into either scenario 2 or 3, and for both, it says to create a dedicated profile for external registrations and put them on port 5090, which works. However, is there no other way to solve this problem that doesn't require the use of an additional profile on port 5090 but also doesn't cut off internally registered calls after 30 seconds? On Asterisk, there's no need to open a second port to register external phones. What's different about Freeswitch? Also, I don't know what role these play, but I also get these errors: [WARNING] switch_core_media.c:1282 Asynchronous PTIME not supported, changing our end from 0 to 20 at seemingly random times ...and.... [INFO] switch_nat.c:590 NAT port mapping disabled when I make a call from internally or externally registered softphone to external number. Thank you.

4 9

Bind to multiple ports with Sofia
by Emrah 07 May '13

07 May '13

Hi guys, I need to bind to multiple ports with Sofia. If I start one profile per port, I cannot get a consistent registration table working. A domain can only be aliased to one profile if my understanding is correct. If you have a working set up, I'd appreciate some advice. Thanks, Emrah

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