October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

issue with mailer_app.php
by Yuriy Nasida 30 Oct '12

30 Oct '12

Hi guys, My mailer_app.php stops working. it is from http://wiki.freeswitch.org/wiki/PHP_email I try to uderstand how to debug this. Fs_cli doesn't say anything usefull. Any help welcome.Thanks.

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Sangoma Vega 50 FXO
by ocset 30 Oct '12

30 Oct '12

Hi Is anyone using/tried a Sangoma Vega 50 FXO gateway with FreeSWITCH? I have used the Grandstream GXW4104 FXO gateway before and I am thinking of switching to the Sangoma as they seem to be more active in FreeSWITCH support and development. Thanks in advance O.

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mod_freetdm problem
by mahesh katta 30 Oct '12

30 Oct '12

Hi List, I have getting problem in EXAWEALTH server, we have installed our telephony software FreeSWITCH for IVR. The problem I am facing when I reboot the server module mod_freetdm is not loading on boot. Please find attachments which is configure for mod_freetdm files. Please find in freeswitch.log file bellow message. "FreeTDM global configuration failed!" switch_loadable_module.c:1309 Error Loading module /usr/lib/freeswitch/mod_freetdm.so **Module load routine returned an error** Kindly suggest me to resolve the issue. Thanking you. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 222, Arunvihar,Sector-28, Noida 201301 GSM +91.99993 45699 | Phone +91.12.0431.0581 Web http://www.buzzworks.com

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How to mix audio in FreeSWITCH
by Khue Nguyen Minh 30 Oct '12

30 Oct '12

Hi all, I want create a js scripts to execute a simple karaoke application. When user make call to my system, js scripts will play a audio file and user can sing. The js scripts will record the song and mix it with audio that is playing. Now, I can create a js to play audio, record call but I don't know how I can mix audio. Please help me. Thanks & Best regards, Khue.

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B-leg Call-ID
by DJB International 30 Oct '12

30 Oct '12

I have only used and enabled a leg parameter in mod_cdr_csv. Is there a way to get the sip_call_id variable from b-leg in mod_cdr_csv without enabling the b legs parameter; something like bleg_sip_call_id ? Thank you.

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Re: [Freeswitch-users] reloading mod_hash...stopping freeswitch
by Peter Olsson 30 Oct '12

30 Oct '12

Read the link I provided, it will tell you what you need to provide when submitting to Jira. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För ram Skickat: den 30 oktober 2012 09:05 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] reloading mod_hash...stopping freeswitch Hi i never submitted any JIRA can some one assists me what all need to submit. On Tue, Oct 30, 2012 at 12:06 PM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: Please report to Jira. http://wiki.freeswitch.org/wiki/Reporting_Bugs /Peter ________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] för ram [talk2ram(a)gmail.com<mailto:talk2ram@gmail.com>] Skickat: den 30 oktober 2012 07:06 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] reloading mod_hash...stopping freeswitch Hi i am having FreeSWITCH Version 1.3.0+git~20121023T150615Z~2246b84554 (git 2246b84 2012-10-23 15:06:15Z) in fs_cli, when the call was on going reload mod_hash freeswitch stoped ? why ? any idea ? Ram _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@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<mailto:FreeSWITCH-users@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:508f873332764633510977!

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Event Consumers
by Nicholas Blasgen 30 Oct '12

30 Oct '12

I have an Event Consumer somewhere that isn't consuming enough events or isn't being alerted to those events. Is there any way to get Freeswitch to dump the list of queued events? switch_event.c:XXX Event queue is full! If I could figure out what pending events there were, I could optimize the function. Nicholas Blasgen Developer Predictive Dialer Limited +1 (724) 252-7436 (cell) Skype: nblasgen

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reloading mod_hash...stopping freeswitch
by ram 30 Oct '12

30 Oct '12

Hi i am having FreeSWITCH Version 1.3.0+git~20121023T150615Z~2246b84554 (git 2246b84 2012-10-23 15:06:15Z) in fs_cli, when the call was on going reload mod_hash freeswitch stoped ? why ? any idea ? Ram

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Question re. load balancing and web-based interface
by Caresolutions 30 Oct '12

30 Oct '12

Hello all I'm new on this list and rather green when it comes to IP telephony in general. During the last year or so I have been managing an existing setup with an Asterisk 1.4 server and a web-based user interface. The user interface is geared towards small medical clinics (typically 2 - 10 users in each) and we currently have in the region of 20 clinics. This setup has been running for 4 years or so. My employees want to expand our client base into the thousands based on the interface which has a lot of functionality. My problem is that our current setup does not allow for load-balancing of the Asterisk server since all the code is written based on the assumption that there is only one server. Our current short-term plan is simply to duplicate the setup if the load on our current Asterisk server gets too big. That way some of our clients would use setup A, others would use setup B etc. This is obviously not optimal and doesn't really give a decent level of fault tolerance. Our web interface allows users to create and manage voice messages, menus, queues, to transfer calls, view callers in queue etc. All sound files can be created through the web interface by utilising the user's phone. Rewriting the code behind the user interface so that it can talk to more than one telephony server in a load-balancing setup is presumably the way to go. This can be done using Asterisk, and I presume, also using FreeSWITCH. But which would be easier? The Asterisk solution would require quite a lot of coding. The problems I can foresee is mostly managing queues and the location of user-created sound files. My preferred solution would be a collection of load-balancing IP servers that presented a unified interface for me to program against, but that may not be possible? I am very much intrigued by the claims FreeSWITCH makes regarding performance, load balancing and fault tolerance. Using XML or writing code is no problem, but obviously I want the best solution with minimum coding effort J I'd love to hear what opinions and pointers you could give me, thanks in advance Binni Care Solutions Brettevillesgade 14 9000 Aalborg Telefon: 8832 1600 Mobil: 3020 0868 Fax: 7262 3921 <http://www.netklinik.dk/> www.netklinik.dk, <http://www.gangweb.dk/> www.gangweb.dk, <http://www.caresolutions.dk/> www.caresolutions.dk <http://caresolutions.dk/> Hjem

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Recording calls with a beep
by Robert Stevens 30 Oct '12

30 Oct '12

Hi All, I have been searching the wiki for a good way to insert a beep every 10 seconds into a call that is being recorded (I plan on recording all calls to Comcast from here on out). This is my dialplan: <extension name="CallCentric.18002662278" > <condition field="destination_number" expression="18002662278" > <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="sip_h_X-Tag=" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" /> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" /> <action application="set" data="inherit_codec=true" /> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}" /> <action application="set" data="RECORD_COPYRIGHT=(c) 2012" /> <action application="set" data="RECORD_ARTIST=FreeSwitch" /> <action application="set" data="RECORD_COMMENT=FreeSwitch" /> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}" /> <action application="set" data="RECORD_STEREO=true" /> <action application="displace_session" data="tone_stream://%(100,15000,800);loops=-1" /> <action application="record_session" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav" /> <action application="bridge" data="sofia/gateway/CallCentric/18002662278" /> </condition> </extension> I would also like to playback a tone so they know I'm recording them (for legal reasons). Per the wiki, the displace_session does indeed play back a tone every 15 seconds. But I'm afraid it never bridges the call.. It's probably something stupid, but what am I doing wrong here? I hear the intial beep after dialing the number but no early media with dead air on the line. What is the best way to insert a tone every x seconds while both legs of the call are in progress and so that each party hears the tone? Also, when testing this without tone_stream, the recording works (horaah) but the recording itself is really ssssssssllllooowwwwww. I am still working on that one, its probably in the wiki, i'm looking now, but if anyone has a suggestion let me know. Best regards, Robert Stevens

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