March 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to add username in SIP packet OPTIONS
by "Гришин В.С." 11 Jun '13

11 Jun '13

My gataway profile <include> <gateway name="SIP_M200"> <param name="username" value="USERNAME_STRING"/> <param name="realm" value="172.30.0.2"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="5"/> </gateway> </include> SIP packet which is accepted by the remote UA Thu Nov 29 16:38:24 2012 RECEIVE 15:5060 <-- 172.30.0.1:5080 OPTIONS sip:172.30.0.2;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.30.0.1:5080;rport;branch=z9hG4bK6DKm54N817tUS Max-Forwards: 70 *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ To: <sip:172.30.0.2>* Call-ID: 79818268-b4c4-1230-60ac-00d0b7ded66a CSeq: 36758947 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20120920T220849Z~f718a5e8e6 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 but in reply arrives nothing... the developer points to that that in a SIP packet there is no field username in *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ <----in this string no username To: <sip:172.30.0.2> <---- and in this string no username* it is necessary to make as From: "USERNAME_STRING" <sip:USERNAME_STRING@172.30.0.1>;tag=9DcB7gSNZZ41j What it is necessary to change or add in a gateway profile (or other place) for this purpose? Vladislav Grishin

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Setting up callbacks
by Alex Lake 07 Jun '13

07 Jun '13

I was wondering where's a good place to find some examples of how, by sending the right commands to the event_socket, I could have Freeswitch establish callbacks for me. Essentially there are a couple of different termination types - PSTN (via a gateway) and internally registered SIP accounts I would wish to be able to set up A->B (and maybe small conferences) using these types of destination in any combination. I've looked through the event_socket pages and the "originate" syntax, but would like to put together something a little more "idiot-friendly" so am looking around for precedents/tips...

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Best Installation MOD_GSMOPEN
by Josue Diaz Cruz 29 May '13

29 May '13

Good Afternoon. I beg your pardon if my question is really from a newbee. I am trying to install gsmopen in a system. I did it with CentOS 6 X64, Debian 6, Ubuntu 12.10. Can some one provide me the best way to do it. Cause i allways have problems in one point of the installation. Depending the distro could it be one or other. I was thinking that could be nice to give a complete requierements (libs) to install before do anything. If some one can provide me a easy, simple and not troubles, could be nice: * System and version * List of requierements. * Better steps to avoid issues with libraries and compile. What i tried: Centos 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues when try to load mod_gsmopen.so) Debian 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues with the ttyUSB port) Ubuntu 12.10 --- Huawey E169 unlocked and voice active. (Issues with the ttyUSB port) E169 works perfectly. I have a mobile partner with voicecall and i test it calling my land line without issues. Thank you very much for your consideration.

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FreeSWITCH 1.0.X.... Really people?
by Ken Rice 29 May '13

29 May '13

So I have noticed lately that tons of people are still trying to install and run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). This is a *VERY* bad idea... These old versions of FreeSWITCH have known vulnerabilities in them and are no longer supported by FreeSWITCH.org period. The 1.2 branch has been released for quite a while now, and is making excellent progress in the terms of both stability and reliability. So now, if you post a Jira for help on FreeSWITCH 1.0 the response you will get a closed bug with the comments to update and try again. If you post to the mailing list you will get told to update and try again If you ask on irc you¹ll get told to update and try again (after getting laughed at and made fun of) K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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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.

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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

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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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ESL Client Library Managed wrappers (Windows)
by Dave R. Kompel 04 May '13

04 May '13

I just spent the good part of this wekend trying to get them compilied, tested and working, it's been over 2 years since I had to build them, and it's a bitch. To save ppl time, I put the binaries up, and made them available if anyone needs them. You can grab them at: http://download.drknetworking.com/eslmanaged.zip --Dave

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