May 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Music On Hold
by Chia-Yen Wu 16 Jun '11

16 Jun '11

Hello, I am using freeswitch as an IVR server. I would like to be able to put the call on hold and play music while performing some other tasks in the lua. when the task is complete , the music stop and the IVR continue but i cant find a way to do that, is there any function can do "music on hold" in dialplan ? Thank you

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UUID_Bridge Fails But Reports +OK
by Antonio Teixeira 09 Jun '11

09 Jun '11

Hello List , Good Morning. In my fight to get ESL & Python & Freeswitch all to behave properly i noticed a possible BUG ( need the veterans to confirm). Scenario : Originate & ValetPark () + MOH OR Just Park() , Both show the same problem. Codec G729 I Make another call: originate & park() codec G729 Now Python ESL Inbound Or FS Console : uuid_bridge uuid1 uuid2 +OK uuid Now one of the two things happen : 1) One the call gets connected hurray :) , Audio Perfect , etc. 2) The call gets dropped :( In both cases uuid_bridge reports +OK even in the case the call is dropped. Even if i Park() Both Calls using the dialplan XML ( NO ESL with those SYN/ASYNC Problems) I still get sometimes ( not always) a dropped call could this be related to the use of G729 ? I have lots of available licenses. Regards António Teixeira

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Setup help Conference, 3 way call, FIFO, MOH, record to Shoutcast
by Rev Michael Carbone 09 Jun '11

09 Jun '11

I have things setup and working and am using fusionpbx and am totally new to all this. I am setting things up to have an online call in talk show. What I want to be able to do is this: Have me and a guest on and the call broadcasting to a shoutcast/icecast server (which I have already) and want to be able to broadcast this as MOH, how can I also have people in a FIFO cue listening in on the show and be able to bring them in one at a time. Broken down: 2 of us on a call broadcasting to shoutcast other people call in and are put in cue with conversation as MOH bring people on air one at a time while others are still muted I'm sure this can be done, but how? Please remember I'm new to all this so instructions would be helpful. Thank you, Michael

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Problem receiving calls from E1 to TE122p
by Javier Galaz Jeria 08 Jun '11

08 Jun '11

Hi, I'm using FS git, on Ubuntu 11.04, with Digium TE122p card (from now on srv1), and FS git, on Ubuntu Server 11.04 with Sangoma A101 card (from now on srv2), both currently configured by vendor's scripts. When I make a call from srv1 to srv2, everything goes well, I can hear both ways no problem, however when I try it the other way around (srv2->srv1) the softphone registered to srv2 gets "User Not Available", hardphone registered to srv1 gets INVITE and just after that CANCEL. The FS log: [1] Looking at the logs I can see that after the RINGING goes to EARLY and tries to play ringback tone, but it can't write and it can't read from channel because it's not opened. So it hangs up. [lines 520-528] I've searched the wiki, asked on IRC, but can't solve my problem, anyone who has configured TE122p could give some pointers? This are my config files: -----TE122p----- /etc/dahdi/system.conf loadzone = es defaultzone = es span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 /usr/local/freeswitch/conf/freetdm.conf [general] cpu_monitor => no cpu_monitoring_interval => 1000 cpu_set_alarm_threshold => 80 cpu_reset_alarm_threshold => 70 cpu_alarm_action => warn [span zt TE122p] trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 /usr/local/freswitch/conf/freetdm.conf.xml <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/> <param name="sip_headers" value="true"/> </settings> <libpri_spans> <span name="TE122p"> <param name="node" value="network"/> <param name="switch" value="euroisdn"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </libpri_spans> </configuration> ------A101----- /etc/wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 2 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = NCRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 430 LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS Blue Alarm and keep line down #wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode #wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode TDMV_HW_DTMF = YES # YES: receive dtmf events from hardware TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events from hardware HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation enabled with nlp (default) # OCT_SPEECH: improves software tone detection by disabling NLP (echo possible) # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions. HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled) HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the line - could break fax HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo cancelation HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone detection (possible echo) HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal [w1g1] ACTIVE_CH = ALL TDMV_HWEC = YES MTU = 80 /usr/local/freeswitch/conf/freetdm.conf [span wanpipe wp1] trunk_type => e1 group=1 b-channel => 1:1-15 b-channel => 1:17-31 d-channel => 1:16 /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml <configuration name="freetdm.conf" description="Freetdm Configuration"> <settings> <param name="debug" value="0"/>    </settings> <config_profiles> <profile name="my_pri_te_1"> <param name="switchtype" value="euroisdn" /> <param name="interface" value="pri_cpe"/> </profile> </config_profiles> <sangoma_pri_spans> <span name="wp1" cfgprofile="my_pri_te_1"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </sangoma_pri_spans> </configuration> How can I fix this? Any help would be greatly appreciated. If you need more info just tell me. [1] http://pastebin.freeswitch.org/16397 Best Regards -j-

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how to use mod_mp4
by pooja 07 Jun '11

07 Jun '11

has anyone used mod_mp4 to play a video. if i use play_mp4 it says invalid application play_mp4. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-to-use-mod-mp4-tp6410226p… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Fwd: Major deployment of outbound FAX on latest version of Freeswitch question
by Steve Underwood 06 Jun '11

06 Jun '11

On 05/12/2011 05:21 PM, Juan Antonio Ibañez Santorum wrote: > Hello Steve! > > I would like to ask you one question about fax system. You told > that using FS+mod_spandsp could be gotten successful faxing rates over > 99%. Looking at your experience, is 99% nearer to 100% or to 99%? > Which errors do you find usually on that 1%? > You can run tests all day in a lab without failure. The real test is how well things work against a wide diversity of machines in the real world. This http://www.soft-switch.org/spandsp-soft-fax-performance.html says something about our testing, and what is achievable. Its quite a lot of work to get real reliability numbers. You get a lot of call failures on a typical public FAX server due to things like wrong numbers, voice calls into a FAX port, people purposefully dropping half completed FAX calls, and buggy FAX machines doing weird things. Don't ignore the last one. There are famous make machines with serious bugs. You need to manually check the failed calls one by one to get real reliability figures. In the end there are always a few calls where you are never sure who's fault the failure is. All I can say with hand on heart is the number of unexplainable failures is well below 1% Regards, Steve

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collectInput does not call callback function
by Kamen 04 Jun '11

04 Jun '11

Hi folks! I seem to have an odd problem, since it works for all others. From my java script I try to collect user input: session.collectInput( mycb, "digits", 7000, 0); where mycb is a regular function collecting user input: function mycb( session, type, obj, arg ) { console_log( "info", "collect digits \n" ); try { if ( type == "digits" ) { console_log( "info", "digit: "+obj.digit+"\n" ); if ( obj.digit == "#" ) { console_log( "info", "detected pound sign.\n" ); exit = true; return( false ); } dtmf.digits += obj.digit; if ( dtmf.digits.length >= digitmaxlength ) { exit = true; return( false ); } } } catch (e) { console_log( "err", e+"\n" ); } return( true ); } //end function mycb The very first line is a console info message, so I would know the function is actually called. It never happens. Right before collectInput call I have a message playing a prompt, which I can hear. So workflow definitely goes into the collectInput function, and then after 7 seconds it continues with dtmf.digits.length=0 even though I punched in the numbers, of course. Thats another proof the callback function never was called. Has anyone experienced this problem before? I run FS in Windows. Appreciate your help. Regards, Sergei Kamen -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/collectInput-does-not-call-ca… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Transferring to a cell phone
by Yungwei Chen 03 Jun '11

03 Jun '11

Hi, I installed freeswitch from a recent snapshot on CentOS 5. I configured a gateway for my SIP provider in /usr/loca/freeswitch/conf/sip_profiles/external/my.xml, and I can make an outbound call to my cell phone using the following javascript snippet. var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234"); Now I am trying to make a call from a land-line to a javascript application in freeswich, which plays a recording and then transfers the call to a cell phone. But transferring the call to a cell phone doesn't work for some reason. Here's how I transfer the call to a cell phone in javascript: session.execute("transfer", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234"); According to freeswitch.log, the problem seems to be that freeswitch cannot find a dialplan that matches the destiantion number, {ignore_early_media=true}sofia/gateway/sip_provider/1231231234. What am I missing here? Thanks.

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Any way to limit the ability to create conference rooms on demand?
by André Rosowski 03 Jun '11

03 Jun '11

Hi there, I'm using Bigbluebutton (online conferencing) with Freeswitch so that users can call a number and then have to enter the "voicebridge" number to be put into a voice conference. The problem, however, is that if a user dialed the wrong "voicebridge" number he will be put into a new conference thus creating one on demand as stated in the describtion of "mod_conference". Is there any way to manually create conference rooms and make them static...not allowing for new conference "rooms" to be created on demand. default.xml in dialplan: <extension name="phone conference"> <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$"> <action application="conference" data="${pin}@wideband"/> </condition> </extension> Thanks for your help : ) Regards, Realdoe -- Omigos Labs UG (haftungsbeschränkt) - Meisenstraße 96 - 33607 Bielefeld Fon: +49 521 2997 200 - Fax: +49 521 2997 101 info(a)omigos.de - www.omigos.de Geschäftsführer: BSc André Rosowski, Benjamin Bittner, Julian Klima Registergericht: Amtsgericht Bielefeld - Registernummer: HRB 40255 USt-IdNr.: 305/5860/1585

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Call For Help: FreeSWITCH Cookbook
by Michael Collins 03 Jun '11

03 Jun '11

Hello FreeSWITCHers! We have a need for some assistance with a few of our recipes for the cookbook. If you are able to write technical documentation in English and have first-hand knowledge of these topics then you might be in a position to help: CDRs: Parsing XML CDRs Handling A and B leg CDRs Using a web server to handle XML CDRs Event Socket: Inbound ESL connections (basic how-to) Outbound ESL connections (basic how-to) Launch an outbound call with inbound event socket & ESL Handle inbound call with socket app & ESL Reacting to events (events for "my call" vs. system-wide events, etc.) Misc: Presence for BLF and SLA Presence for FIFO agent status If you think you've got the chops to write one or more recipes for the cookbook then email me off list and we'll discuss the specifics. I will get you all the information you need to get started. Thanks! -Michael

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