April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Have someone zrtp with correct sas ?
by J.B. BERLIN 29 Aug '13

29 Aug '13

Dear users, I try run FS last git build from today 19. June 2011 with ZRTP. Zfone show me green but the sas are everytime different! Tested with 2x Notebooks on Fedora 14 Twinkle(via yum) & with two desktops with Win7 x64 / Zfone 0.92.build.219 and X-Lite on both sides. FS is build with libzrtp-0.81.514 zrtp.lua is enabled ! global_setvar zrtp_secure_media=true" is set Or have someone a real good step by step guide ? Many thanks and best regards Jacek

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Faxing problems if B leg offers T.38
by Rob Morin 04 Jul '13

04 Jul '13

Im having problems faxing if the B leg offers T.38. Several scenarios, but always the same result. First, the architecture FAX à ATA (T.38 optional) à FS à Carrier à Destination (Possibly another carrier (PSTN) and the fax machine) On my ATA, T.38 is enabled (Auto Detect). If the destination supports T.38, the Carrier will offer it to Freeswitch. If the Destination doesnt support T.38, the B leg doesnt offer it and the faxes go through. Ive set Freeswitch as: <action application="set" data="fax_enable_t38=true"/> <action application="export" data="t38_passthru=true"/> When thats the case, and the B leg offers T.38, Freeswitch passes through the offer and the ATA responds, 3 times. But the response never gets sent to the B leg, so it terminates the call after about 30 seconds. When I set Freeswitch as: <action application="set" data="fax_enable_t38=false"/> The B leg still offers T.38 if its other side is capable. Freeswitch doesnt pass the offer through and the B leg terminates the call after about 30 seconds. Im running on a recent git ( FreeSWITCH Version 1.0.head (git-86d757d 2011-05-17 22-51-47 -0500) ). Ive had no success attempting to configure FS as a gateway. Frankly, T.38 passthrough has had the most success, mostly when the other ends arent T.38 capable and it stays in G711u. Is there something else I need to do to enable T.38 in passthrough mode? Thank you, Rob Morin PS I can provide tcpdumps of this, or whatever else is necessary.

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Dynamic specify the outbound GW within source code
by fieldpeak 27 Jun '13

27 Jun '13

Dear friends, i have FS for PSTN outbound call using below dial plan, <extension name="CallToPSTN" > <condition field="destination_number" expression="^(0\d+)$" > <action application="bridge" data="sofia/internal/$1(a)1.2.3.4" /> </condition> </extension> While, now i need dynamically specify the outbound GW’s IP address according to the return result of the external command before routing in the source code , e.g. if the external command return FS the IP address of OB GW 6.7.8.9, then <action application="bridge" data="sofia/internal/$1(a)6.7.8.9" /> however, i don't know which function i should call within the source code to realize it, could anybody help to advise, P.S. i know there is existing module “mod_xml_curl” can realize similar function, however, I could not use it for this case… thanks a lot! Regards, Charles

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voicemail_say_phone_number
by Yuriy Nasida 24 Jun '13

24 Jun '13

Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise.Thanks.

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Anyone able to get freeswitch to register a gateway over TLS or TCP? I can't get it to not send UDP packets for reg.
by Mitch Capper 14 Jun '13

14 Jun '13

I am trying to get freeswitch to register with another freeswitch server (on TLS port 5061) however default the config settings I believe should set it, it always trys to register over UDP (after registration or with registration set to false however it does send TCP then for outbound calls). If anyone has a gateway config that actually registers over TCP or TLS that would be great. The client gateway/profile config at: http://pastebin.freeswitch.org/15383is what i believe should work but isn't I have tried setting contact-params on the gateway to tport=tls tport=tcp tport=tcp;transport=tls and transport=tls without any luck. Anyway even TCP would be helpful as it would allow me to trace better how the sip registration decides to use TCP or UDP. I tried digging into libsofia but its a bit hard to trace and understand with its generic event structure and tagging (plus the documentation isn't the most rebust on registration options). Thanks ~Mitch

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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

Check if UA is still there?
by Peter Steinbach 03 May '13

03 May '13

Hello, is there a way to check if a UA is still available? The reason is, that sometimes a channel hangs and is not being terminated because the UA was disconnected due to a network failure. As the media is outside Freeswitch (bypass media), it is difficult for us to determine, if the call is still active. Meanwhile the B leg to PSTN is also not hungup and produces costs. Is there a way Freeswitch can detect this? Or is there a way Freeswitch may send a SIP message to the UA by an external process and wait for an OK which can be captured e.g. by event socket interface? An help is welcome. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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MPL v2
by Puskás Zsolt 12 Apr '13

12 Apr '13

Hi. A new version of the MPL license just came out ( http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better than the last one? I'm trying to find the differences but this kind of text is to hard for me :) Any plans for changing FreeSWITCH license to the new one ? Zsolt

5 6

ooH323 vs mod_h323
by Saeed Ahmed 17 Feb '13

17 Feb '13

Dear FSers, which one is better to use when there are higher number of concurrent calls? *ooH323 *or *mod_h323*. Thanks

9 22

Callgroup keeps ringing after answer - changed from shared extensions
by Mario G 04 Feb '13

04 Feb '13

Been at this for 3 weeks now, hope someone can help.... This started after replacing shared extensions with unique users and updating from an early 2011 git. Ring group via "pointer" works fine as it always has. However, answer via callgroup variable intermittently (30-40%) leaves one or mores phone ringing even though the the call was answered at another extension. The darn thing is that I can't reproduce it myself to get a trace. Any help/suggestions greatly appreciated. Phones are SPA962 (UDP) and Bria on iPad (TCP, same problems) and they all stopped ringing when they shared IDs. Mario G default.xml ring-group has (abbreviated to save space)------------------: <group name="home"> <users> <user id="100" type="pointer"/> <user id="101" type="pointer"/> ..... more users </users> </group> <group name="work"> <users> <user id="200" type="pointer"/> <user id="201" type="pointer"/> </users> </group> </groups> Extension----------------------: <user id="101"> User: <params> <param name="password" value="$${default_password}"/> </params> <variables> <variable name="voicemail_domain" value="xxxxxx.com"/> <variable name="toll_allow" value="domestic,domestic10,international,local"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Front Desk"/> <variable name="effective_caller_id_number" value="101"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>  <variable name="callgroup" value="home"/> </variables> </user> Bridge command-----------------: <extension name="xxphone"> <condition field="destination_number" expression="^2920$"> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>  <action application="execute_extension" data="2925 XML default"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ringback=file_string://${xxsounds}hi-long.wav!${sayname}!tone_stream://${us-ring};loops=-1"/>  <action application="bridge" data="{originate_timeout=17}${group_call(home@${domain_name}+A)}"/>

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