August 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

28 Aug '10

It's interesting the wiki goes to great lengths to explain the difference between an automated attendant and IVR and elsewhere in the wiki in the list of features Automated Attendant is shown, however there is no mention of usage/configuration etc. So I'm left to think that you have to roll your own Automated Attendant using IVR which brings to mind the question "why suggest there are two different things when they appear to be the same?" But my question really is, "is there an Automated Attendant in FS separate from IVR and if so can I get some details?" Thanks, Jim.

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Skype help
by Sameer Khan 28 Aug '10

28 Aug '10

hi all i am trying to use multiple instances of same skype user, i want to run 20 instances skype version is the latest downloaded from skype website, i couldnot find skype 2.0.72 FS is the latest git my db files are located in /root/multi/01/skypeuser similarly /root/multi/02/skypeuser my multi.sh is: #start the fake X server on a given port /usr/bin/Xvfb :101 -screen 0 800x600x16 -nolisten tcp -ac & sleep 3 # start a Skype client instance that will connect to the X server above, and will login to the Skype network using the 'username password' you send to it on stdin. Here passwd5 would be the password and user5 the username su root -c "/bin/echo 'skypeuser passwd1'| DISPLAY=:101 /usr/bin/skype --dbpath=/root/multi/01 --pipelogin &" exit 0 but skype is unable to login... this user can not come online.. am i making any mistake in dbpath or somewhere else???? also when i kill the Xvfb it says FreeFontPath: FPE "unix/:7100" refcount is 2, should be 1; fixing. thanks

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playback options
by broken dash 28 Aug '10

28 Aug '10

I'm pulling shout cast streams into freeswitch using the playback action and I'm wondering if there is a variable like playback_delimeter that I could set within my dialplan/scripts that would essentially load up the stream and chop off a configurable amount of time before it essentially bridges up the audio to the caller? or perhaps there is a dialplan routine that could do this for me? Brian

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weird bootstrap error
by lloyd thomas 28 Aug '10

28 Aug '10

Got a wierd error (i think) running bootstrap.sh. It stop at the following line and did not return to the linus prompt. /usr/src/freeswitch# automake: compiling `check_dlopen_sofia.c' with per-target flags requires `AM_PROG_CC_C_O' in `configure.ac' tests/Makefile.am:37: while processing program `check_dlopen_sofia' Is this normal?

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Become a FreeSWITCH NINJA - Signup for FreeSWITCH Training in New York City!
by Darren Schreiber 27 Aug '10

27 Aug '10

Hi folks, After a very successful 1st training, I'm pleased to announce our second official FreeSWITCH Training in New York City! The training is a 3-day bootcamp where we'll dive deep into FreeSWITCH and all the various goodies FreeSWITCH has to offer. Scheduled from October 13th-15th in downtown Manhattan, you'll learn everything you've ever wanted to know to master FreeSWITCH, including: * Understanding configuration files and the default configuration * Call authentication and routing basics * Integration modules (mod_skypiax, mod_dingaling for Skype/GTalk/XMPP integration) * the FreeSWITCH event system * Load balancing and high availability * FreeSWITCH Internals * How to debug and troubleshoot FreeSWITCH * Building Custom C Modules * Advanced Modules Registration is open now and group discounts are available. Please contact me for more information! Hope to see you in New York! For more information, visit http://www.voipkb.com/ Sincerely, Darren Schreiber the 2600hz Project www.2600hz.org<http://www.2600hz.org>

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long DTMF
by Tihomir Culjaga 27 Aug '10

27 Aug '10

hello, i just got a question. Is it possible somehow to achieve a DTMF collect cancel feature. quite hard to explain :)) lets suppose you have an extension that collects some digits ... e.g. 123456789 .. and in the middle you realized you entered a wrong digit and u want to correct it. Is there any way you can assign a digit e-g. * or # to interupt te current input and to start over again by lets say transfering back to the extension collecting digits? Any hint ?

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External caller-number not shown on forward to external number
by Peter Waldheim 27 Aug '10

27 Aug '10

Hi all, In a diaplan I'm forwarding a call to an external (PRI) number when it fails to reach my SIP-Client: <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>  <action application="set" data="outbound_caller_id_number=${caller_id_number}"/> <action application="set" data="outbound_caller_id_name=${caller_id_name}"/> <action application="transfer" data="#external-number# XML default"/>  Then in that dialplan: <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="openzap/1/a/$1"/>  So far this works just fine but one thing: the external party only sees the original caller-id if it is an internal one. If the caller was external, I only see the default caller-number of my installation. Ideally I'd like to always see the original caller id - especially if it is an external number. Can anybody please give me a hint whether I'm missing something or this might be some call-spoof protection in freeswitch or openzap or libpri or my PRI provider? Thanks a lot Peter ________________________________ Framesoft AG Software Applications Sumpfstrasse 15 6301 Zug Switzerland Handelsregister des Kantons Zug: CH-170.3.022.876-2 Pr?sident & Vorsitzender der Gesch?ftsleitung: Toralf Dittmann Framesoft AG Software Applications Reuterweg 49 D-60323 Frankfurt am Main Germany HRB 34142 Vorsitzender des Aufsichtsrates: Toralf Dittmann Vorstand: Jens Saarholz Framesoft Ltd. Business Address: 60 Lombard Street London EC3V9EA, UK Registered Office: Hackwood Secretaries Limited 1 silk Street London EC2Y 8HQ, UK Company Number: 4055017 Directors: Toralf Dittmann, Volker Weber Secretary: Volker Weber Confidentiality Notice: The information contained in this e-mail is intended for the named recipient(s) only. It may contain privileged and confidential information, and if you are not the addressee or the person responsible for delivering this to the addressee, you may not copy, distribute or take action in reliance on it. If you have received this e-mail in error, please notify us immediately by returning the original message to the sender by e-mail and delete this message. Thank you for your cooperation.

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New wiki page
by Jim 27 Aug '10

27 Aug '10

For what it's worth, I documented my experience with getting a Sangoma A200 card working on FreeSWITCH. I have no idea if it's the right way but it works sort of. http://wiki.freeswitch.org/wiki/Sangoma_A200 Jim.

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voicemail profile dbname
by François Delawarde 27 Aug '10

27 Aug '10

Hello, Is there any way to setup the name of the internal database for voicemail? The default seem to use one database per voicemail profile such as: voicemail_<profile_name>.db I would like to be able to use multiple voicemail profiles with the same users if possible, and share the database between all voicemail profiles. Are the vm profiles made to be able to work that way? Thanks, François.

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Serious and urgent problem with DTMF! Please help...
by Dennis 27 Aug '10

27 Aug '10

hi there, at the moment we have a serious and urgent problem with transmitting dtmf-inputs in the right way from one endpoint through fs to another endpoint. we are not very shure, if the problem is on our or on the carriers side. digits send through fs from one side to another are not received correctly. is there someone who could help us (quite quickly)? we are receiving the dtmf-signals on fs as RFC-2833. our fs-setting are the following: <param name="rfc2833-pt" value="101"/>  <param name="dtmf-type" value="info"/> <param name="dtmf-duration" value="100"/> - we are receiving a call and we connect to another side through fs. - the caller enters the following dtmf-digits: 987654321. - fs receives the dtmf-inputs absolutely correctly !!! - the target receives the following dtmf-digits: 98877665543321 it is important to say, that the call comes over the "normal phone-line" (or what ever a phonecall is called, which does not come over voip - landline?). per VOIP (cirpack) our fs-servers receive this call/dtmf-inputs and then we send the call to the target over the "normal phone-line". does anybody have a clue where our problems might be? what could we do to find out more about this probleme? where could we debug? thanks a lot dennis

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