November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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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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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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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

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Spandsp modems as non-root?
by Alex Crow 06 Apr '13

06 Apr '13

Hi, Has anyone figured out how to get the spandsp emulated modems running when Freeswitch is running as a non-root user yet? I don't even get the /dev/pts/pts* devices added either on debian or Ubuntu. Thanks Alex

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ERROR! unable to open ZRTP cache file
by Vbvbrj 22 Mar '13

22 Mar '13

Hello. Regularly in FS log I see: 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: Storing ZRTP Cache... 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: Storing ZRTP cache to <... 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: ERROR! unable to open ZRTP cache file <. 09:03:54.044779 [DEBUG] switch_rtp.c:915 [ zrtp cache]: Storing ZRTP Cache - DONE. 09:03:54.044779 [DEBUG] switch_rtp.c:776 Saving ZRTP cache: OK Why is this logged, and why an error? There is no file name to search for permission or access. Note the missing ">" where file name must be. -- Mimiko desu.

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Playing of mms stream by means of FS
by Yuriy Nasida 05 Mar '13

05 Mar '13

Hi guys What the best way for the playing of mms stream like this mms://live.rfn.ru/rcult_64 by means of FS ?I try to use mod_shout but without success. Do I have to use mplayer or vlc for this ?I saw this in wiki http://wiki.freeswitch.org/wiki/Mod_vlc But... probably there is the more better way ? Please adviseThanks.

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Re: [Freeswitch-users] Concatenated multiple playback types
by Peter Olsson 28 Feb '13

28 Feb '13

I don't believe this is possible. If you record 5 seconds of silence into a file you might be able to use file_string:// to play them both. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Emrah Skickat: den 22 augusti 2012 16:24 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Concatenated multiple playback types Hello there, I am trying to add some silence before the alone sound file in my conference.cnf.xml and can't get it to work. Here is what I tried: <param name="alone-sound" value="silence_stream://5000!file_string://conference/conf-alone.wav"/> <param name="alone-sound" value="silence_stream://5000&file_string://conference/conf-alone.wav"/> <param name="alone-sound" value="silence_stream://5000|file_string://conference/conf-alone.wav"/> <param name="alone-sound" value="silence_stream://5000;file_string://conference/conf-alone.wav"/> Any idea on how to do this? Thanks! Emrah _________________________________________________________________________ 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 !DSPAM:5034e9a632761303016055!

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