May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Lua creating multiple session's
by Jon Schøpzinsky 23 May '13

23 May '13

Hello List, Im trying to do dial multiple destination through lua, on a single incoming call. I do know that i could do a simple session:execute("bridge", "dst1,dst2,dst3") but I need to do it in individual session, for processing I need to do in a later version of the lua script. I loop through the destinations that needs to be called, creating a new session for each destination, and storing that in an array. Firstly, it seems as freeswitch.Session doesn't reply right away, but waits for early-media. Thats ok though, but makes dialling mobile devices a rather long wait. The problem is, that when I do the second freeswitch.Session, it seems to hold up further lua processing, until the last created call is answered. Is this how its supposed to work. Heres my current code: local legs = {} for key, dev in pairs(dstDevices) do freeswitch.consoleLog("info", "Lets call " .. dev.username .. " with tech " .. dev.devicetech .. "\n") if dev.devicetech == "1" then freeswitch.consoleLog("info", "calling " .. dev.username .. " on uasbc\n") legs[key] = freeswitch.Session("sofia/gateway/uasbc01/" .. dev.username); freeswitch.consoleLog("info", "called " .. dev.username .. " on uasbc\n") elseif dev.devicetech == "2" then freeswitch.consoleLog("info", "calling " .. dev.username .. " on ccsbc, mvno\n") legs[key] = freeswitch.Session("{origination_caller_id_name=+xxxxxxxx,origination_calle r_id_number=+xxxxxxxxx}sofia/gateway/ccsbc01/+xx" .. string.match(dev.username, "^mvno_(.+)")) freeswitch.consoleLog("info", "called " .. dev.username .. " on ccsbc, mvno\n") end End freeswitch.consoleLog("info", "Enter loop now\n") It doesn't reach the last consoleLog until the last call is answered. Venlig hilsen/kind regards Jon Leren Schøpzinsky

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Re: [Freeswitch-users] Conference questions
by Matthew Cordes 23 May '13

23 May '13

Hi Mike, Thanks for the quick response. Here are a few more questions / comments below: > 1. How do I translate conference cli commands (e.g. "conference xxx mute > yyy") into a dialplan xml actions? > > You don't really. Why exactly would you want to do this? > In some situations I'd like to mute all users other than the moderator from when they join for the duration of the conference. I thought the right way to do this would be via an action in my dialplan as new callers join my conference. What would you recommend? Thanks, Mod_xml_curl looks straightforward enough. 4. I'd like to have some private conferences that only certain callers can join. I'll know the callers' ids before hand. I'm assuming a reasonable way to handle is with user groups. How would I test if a user is a member of a particular group? If your doing dialplan via mod_xml_curl you can do all that validation in > your dialplan lookup. > Can you point me to a resource regarding how I would do this validation? I know how to define users and groups. It's not clear to me how to test for membership in a group or how that would be written (although I'm assuming it would be a condition in the dialplan, correct?). -Matt

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Cannot ring extension from DID
by Mike Hendrie 23 May '13

23 May '13

*==========================* *Server:* Ubuntu 12.04 *Version:* FreeSWITCH Version 1.5.1 *Enabled:* Multi-Tenant *Extensions:* 1000 and 1001 *Connection:* freeswitch server NAT to provider *Connection: *extensions are on the same LAN as fs server *DID:* 1 DID ========================== *Can-Do: * - I can dial extension to extension (1000 to 1001) and the phone rings and can talk. - I can test with extension 5000 and listen to the echo test. - I can dial the DID to the Freeswitch server. The server sends the the call to voicemail, 1000, when it should ring the phone. *Cannot Do:* - I cannot call out of any extension. The phone does not ring. It is silent. - I cannot dial the DID and ring a phone. It only goes to VoiceMail. *Question:* - Why can I not dial the DID and have the extension ring? - Where should I check? Thank you, Mike

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Conference questions
by Matthew Cordes 23 May '13

23 May '13

Hi, I'm new to FreeSwitch and starting to experiment with conferences. I've browsed the wiki and I'm half way through the ebook and I'm running into a few questions I'm hoping I might be able to get some help with. 1. How do I translate conference cli commands (e.g. "conference xxx mute yyy") into a dialplan xml actions? 2. The default behavior of the conference is to disable hold music and allow the participants to speak when there are two or more callers. How might I keep the conference on hold until a particular caller joins? Additionally when this special user leaves I'd like to place the conference back on hold. 3. Where might I find info regarding loading configuration information (particularly the user directory) from an external source (probably a database and probably via mod_python)? 4. I'd like to have some private conferences that only certain callers can join. I'll know the callers' ids before hand. I'm assuming a reasonable way to handle is with user groups. How would I test if a user is a member of a particular group? Thanks, -Matt

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Max. Number of 8 span PRI cards
by Ashish gautam 23 May '13

23 May '13

Hi, I want to ask that is it possible to use 3 eight span PRI cards on a single FS server? I mean is there any limit up to which the system performs optimally on this? Please throw some light on this. Thanks -- Ashish Gautam IVR Developer Nucleus Microsystems (Pvt.) Ltd. +91 8802865008

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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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SDP manipulation and sofia_glue.c (developer help)
by Daniel Ivanov 23 May '13

23 May '13

I am having difficulties forcefully stripping the a:crypto lines from the remote sdp string. At first i was doing <action application="set"><![CDATA[switch_r_sdp=$sdp]]> </action> from the diaplan, but this breaks the proxy media easily, because it doesn't patch the SDP later for glueing. I commented the following lines where the patcher is checking if the sdp string has been set before : void sofia_glue_tech_patch_sdp(private_object_t *tech_pvt) { switch_size_t len; char *p, *q, *pe, *qe; int has_video = 0, has_audio = 0, has_ip = 0; char port_buf[25] = ""; char vport_buf[25] = ""; char *new_sdp; int bad = 0; */** * if (zstr(tech_pvt->local_sdp_str)) {* * return;* * }* **/* And now i got the correct sdp sent to the b-leg(patched for proxy media and nat), but no audio is going either way. Would there be a more clever way to "rip" some lines from the SDP and not break the NAT/Proxy media processing afterwards? Help is much appreciated.

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The number of registered phones
by andpe 23 May '13

23 May '13

Hi What is the possible number of registered phones to FreeSWITCH? I know it may depend on many factors. Where can I find this information? Is 2000 or 3000 users a lot or not? I'm not asking about the performance of connections here. Regards Andy

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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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ANNOUNCEMENT: ClueCon Hotel Registration Available Online
by Michael Collins 22 May '13

22 May '13

The ClueCon team would like to announce that the hotel for ClueCon 2013 may now be booked online. Please visit www.ClueCon.com and click the "Book Your Room Online" button. Once you are at the Hyatt site you can enter in your check in and checkout dates and click the Search button. You will be presented with your choice of "standard king" or "standard double" rooms. Keep in mind that the standard double rooms go fast, so don't delay! Be sure to register <http://www.cluecon.com/register/> for ClueCon right away so that we can plan our various activities for when we're all in Chicago. See you in August! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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