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while in a bridge - is it possible to bridge one leg to another destination
by Ali Pey 27 Jan '14

27 Jan '14

Hello, Say A is bridge to B. I have dind_digit_action set for some in call feature. When the digit is detected: 1- I need to put the A leg on hold 2- Bridge the B leg to another destination I can't use att_xfer as it doesn't have all the features I need. Is it possible to have nested bridge? Is it possible to kill the first bridge and the bridge the calls again later? Right now, I get no speech path on the second bridge and I can't find a way to kill the first bridge, it's always there until one leg hangs up. Is there a way to achieve this? Any suggestions? Thanks, Ali Pey

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Will sqllite on a ramdisk be more performant than postgres with the db on a ramdisk?
by Clinton Goudie-Nice 27 Jan '14

27 Jan '14

I've got a piece of tech that I'm working on where I don't really need to worry all that much about recreating the FS DB on a reboot. With that in mind, I'm thinking about one of two options: 1) Sqllite on a ramdisk. 2) Postgres with the db for FS placed on a ramdisk. What do you think I will get higher performance out of?

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Audio running issue
by Web Developer 27 Jan '14

27 Jan '14

Hello, Why I am getting this error when I originate call. [WARNING] switch_rtp.c:6221 Generating RTP locally but timestamp passthru is configured, disabling... Thanks

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originate and record command help
by sunil more 27 Jan '14

27 Jan '14

Hi All, I am trying originate and bridge a call from the freeswitch CLi usng the below command, originate {ignore_early_media=true}sofia/mypbx.com/7011@192.168.1.142:5060 &bridge([absolute_codec_string=PCMU]sofia/mypbx.com/7012@192.168.1.142:5060) the above command works fine. However I want to know how can I modify the above command to record this call session once this call is bridged. I dont want to use a different command but just pass a command with the originate command. Thanks in advance. -- Sunil More

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FIFO a video call?
by efernandez99＠hushmail.com 27 Jan '14

27 Jan '14

Good evening FS fellows, Is there any way to make video calls (H.264/H.263+) be queued like modfifo does with regular voice calls? Want to setup multiple grandstream GXV3175 auto off hook into a queue where Bria agents pick the video call up without toggling from voice to video. I'm doing it right now with regular bridge extensions but wanted to see if there was a way to put them in some sort of 'queue'. Be cool to even have it do a play_fsv for the 'hold music/video' ?? Thanks, Eric Fernández

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Hangup in CLI
by Alex Lake 27 Jan '14

27 Jan '14

When attempting to start a freeswitch 1.2.0 install, I get a bunch of start up messages, a pause and then "Hangup". What does this mean?!

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Outbound calls via TDM2400P analog card using FreeTDM
by Errol Samuels 27 Jan '14

27 Jan '14

Hello All, I seem to have an issue being able to make outbound calls via a Digium TDM2400P configured with Dahdi and FreeTDM. I searched and followed the instructions on the Freeswitch site as well as Sangoma's site but can't seem to get any further so I am hoping that someone could double check my config and provide some feedback. The test calls are being sent from a registered Jitsi softphone. I am not getting any progress tones when I send the call from FS and the call appears to hang for about 30 secs or so. When I test with Asterisk I get the progress tones so perhaps there is some issue with my config. The full log is available here: http://pastebin.freeswitch.org/21901 freeswitch@internal> ftdm dump 1 1 span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 physical_status: ok physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: UP type: FXO state: DOWN last_state: DOWN txgain: 2.00 rxgain: 3.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE session: (none) **** My configuration is as follows: **** ## /etc/dahdi/system.conf # Span 1: WCTDM/0 "Wildcard TDM2400P" (MASTER) fxsks=1 echocanceller=mg2,1 fxsks=2 echocanceller=mg2,2 . . <snipped> . . fxsks=23 echocanceller=mg2,23 fxsks=24 echocanceller=mg2,24 # Global data loadzone = us defaultzone = us ## /usr/local/freeswitch/conf/freetdm.conf [general] cpu_monitor => no cpu_monitoring_interval => 1000 cpu_set_alarm_threshold => 80 cpu_clear_alarm_threshold => 70 cpu_alarm_action => warn debugdtmf_directory=/var/log/freetdm.log iostats => yes [span zt myDahdiFXO] name => myDahdiFXO trunk_type => fxo analog-start-type => kewl group => grp1 txgain=2.0 rxgain=3.0 fxo-channel => 1-24 ## /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/>  </settings> <analog_spans>  <span name="myDahdiFXO"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="answer-polarity-reverse" value="true"/> <param name="tonegroup" value="us"/> <param name="enable_callerid" value="true"/> <param name="enable-analog-option" value="call-swap"/> </span> </analog_spans> </configuration> ## Dialplan ## /usr/local/freeswitch/conf/dialplan/default/10_Outbound_FXO.xml <include> <extension name="Outbound_via_FreeTDM"> <condition field="${toll_allow}" expression="domestic"/> <condition field="destination_number" expression="^9(\d{8})$"> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="freetdm/myDahdiFXO/a/${destination_number}"/> </condition> </extension> </include> Hope I have given enough info. Thanks in advance Errol

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Re: [Freeswitch-users] trans-coding is not working
by Shahzad Bhatti 27 Jan '14

27 Jan '14

i was to do transcoding on a bridge call not try to negotiate codec only like on legA i have GSM and i have codec pref for outbound as PCMA,PCMU then LegA got response after transcoding with PCMA so i put inbound-late-negotiation to true. Regards Shahzad Bhatti ---------- Forwarded message ---------- > From: "Dahlberg, David" <david.dahlberg(a)fkie.fraunhofer.de> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 24 Jan 2014 14:01:56 +0000 > Subject: Re: [Freeswitch-users] trans-coding is not working > Am Freitag, den 24.01.2014, 16:48 +0500 schrieb Shahzad Bhatti: > > no i want to transcode GSM with PCMA so i > > <param name="inbound-zrtp-passthru" value="false"/> > > <param name="inbound-late-negotiation" value="true"/> > > to enable late negotiation only for trans-coding. > > Please try "inbound-late-negotiation=false" to negotiate both call legs > independently. (see http://wiki.freeswitch.org/wiki/Codec_negotiation) > > -- > David Dahlberg > > Fraunhofer FKIE, Dept. Communication Systems (KOM) | Tel: +49-228-9435-845 > Fraunhoferstr. 20, 53343 Wachtberg, Germany | Fax: +49-228-856277 >

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make current -- NOT
by Sean Devoy 27 Jan '14

27 Jan '14

HI, I realize all the heavy hitters are off partying and in a FreeSwitch frenzy by now, but can someone look at my pastbin log http://pastebin.freeswitch.org/21892 and tell me why I can't "make current"? I am happy it did NOT leaving with a dead server this time! That is truly progress and I am VERY HAPPY about it. I am not a Centos guy (no duh), but Unix boxes are too much cheaper than windows boxes for the same use. Any ideas would be greatly appreciated. We are also entertaining reasonable offers for Administrator(s) to keep our Centos / Freeswitch server up to date and on stable versions. We can handle the FS config. A decent hourly rate or monthly retainer maybe? Thanks, Sean

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(no subject)
by Andrea Coslovich 27 Jan '14

27 Jan '14

Hello, I have an A104 Sangoma board and I'm developing an application based on FreeTDM library. In my application I need to answer to incoming calls as soon as possible, so I've tried to call "ftdm_channel_call_answer(channel)" immediatly after the receiving of the "START event" and I was really displeased when my call didn't rise and I saw a "VETO state change from RING to UP" message in my log. As far as I know the ISDN protocol allows to send a CONNECT message immediatly after receiving a SETUP, so I'm wondering why I cannot call the "ftdm_channel_call_answer(channel)" in a "RING" state. Is this a problem of my software or the normal behaviour of the FreeTDM library? Thanks a lot for any feedback! Kind regards, Andrea Coslovich

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