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New build gives error message for default grammar file??
by mszlazak＠aol.com 25 Feb '09

25 Feb '09

I'm getting this error message trying out the pizza demo in FS 1.0.3: ?"Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks.

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howto originate fs call from webapp (python)
by Alexander de Greiff 24 Feb '09

24 Feb '09

hi all, i come from asterisk an i am new to freeswitch. after my with days with freeswitch i am very excited! but trying to migrate our deployment i have three challenges. one of them is: i need to call freeswitch from a webapp (e.g. python) and pass number1 and number2. i then need freeswitch to call number1. as soon as it is picked up say a short confirmaton text, call number2 and bridge the two. my first approach was to call via xml_rpc like described in the wiki but when i call like server.freeswitch.api("originate","sofia/gategay/gateway1/{number1} &bridge(sofia/gateway/gateway2/{number2})") but in this case both numbers are called in parallel and the first number to pick up gets a ringback tone until the other number picks up. how can i get the sequence described above? thanks for your help alex

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Web-based forum?
by Fred 24 Feb '09

24 Feb '09

Hello Maybe this question has been raised before, but if not: There's so much traffic in this mailing list that I was wondering if adding a web-based forum on the site was in the works? Cheers,

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Recording and outbound rtp
by Dan 24 Feb '09

24 Feb '09

Hi, I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great. The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch. Thanks! Dan-

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file directory.conf.xml
by Ali Al-Rubaie 24 Feb '09

24 Feb '09

Hi, The file directory.conf.xml had been mentioned in the documentation many times but there is not such file in the conf folder. Do you mean default.xml in directory folder? Thanks! --- On Tue, 2/24/09, freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> wrote: From: freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> Subject: Freeswitch-users Digest, Vol 32, Issue 181 To: freeswitch-users(a)lists.freeswitch.org Date: Tuesday, February 24, 2009, 3:34 AM Send Freeswitch-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: SIP dump to DB (kokoska.rokoska) 2. FREESwitch on Windows Server 2003 (Stephen Walker) 3. Re: mod_erlang_event compile problem (Andrew Thompson) 4. Re: FREESwitch on Windows Server 2003 (Carlos Talbot) 5. Re: SIP dump to DB (Joseph Bajin) 6. Re: SIP dump to DB (kokoska.rokoska) 7. Re: mod_portaudio: Do not accept next call after Hangup (Rene Pankratz) 8. Patch for openzap concerning finding a free channel. (Helmut Kuper) ---------------------------------------------------------------------- Message: 1 Date: Mon, 23 Feb 2009 23:32:26 +0100 From: "kokoska.rokoska" <kokoska.rokoska(a)post.cz> Subject: Re: [Freeswitch-users] SIP dump to DB To: freeswitch-users(a)lists.freeswitch.org Message-ID: <49A323FA.8000802(a)post.cz> Content-Type: text/plain; charset=ISO-8859-1 Joseph Bajin napsal(a): > Basically, you are trying to build what Empirix has with their Hammer tool. > Thank you very much, Joseph, for your interest! I have never heard about Empirix (I'll look at it), but what I'm trying to build is something like SER/Kamailio/OpenSIPS sip_trace module. > You can create an application that is basically a mix of tshark and a > database feeder. > You sniff with tshark and going to basically pipe it to another > application that will read the pcap file, parse it, and load it into the > db for you. There are plenty of modules out there that will read pcap > for you. > Thank you once more, Joseph, for suggestion! I think about it - it will be challenge for me to write robust and still fast enough (thousands messages per second) SIP parser + DB feeder :-) Best regards, kokoska.rokoska ------------------------------ Message: 2 Date: Mon, 23 Feb 2009 14:47:13 -0800 From: "Stephen Walker" <swalker(a)SONASEARCH.com> Subject: [Freeswitch-users] FREESwitch on Windows Server 2003 To: <freeswitch-users(a)lists.freeswitch.org> Message-ID: <3B93E0500B57D04CBAE85520B750CFF04CA6CE(a)exchange.sonasearch.com> Content-Type: text/plain; charset="us-ascii" Hello: I have successfully loaded the Windows implementation (SVN 11602 - 02/02/09) from your site and it runs fine. I configured a Linksys SPA 2102 and have acquired dial tone and the '999X' tests work. I have not been able to establish connection with either FreeWorldDialup or Broadvoice as of yet. Which files do I need to edit and what are the proper entries to enable connection to FreeWorldDialup and Broadvoice? Example files and where they reside in the file structure would be very much appreciated. Thank you All the Best, Steve Steve Walker President SONASEARCH, INC 425/883-1984 NOTICE: The information contained in this document is intended by Sonasearch, Inc. or one of its subsidiaries for the use of the named individuals or entities to which it is addressed and may contain information that is privileged or otherwise confidential. It is not intended for transmission to, or receipt by, any individual or entity other than the named addressee (or a person authorized to deliver it to the named addressee) except as otherwise expressly permitted in this document. If you have received this document in error, please destroy it without copying or forwarding it, and notify the sender of the error by calling Sonasearch at (425) 883-1984.

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[ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform
by Andrew Thompson 24 Feb '09

24 Feb '09

Hi, Today I'd like to announce the open-sourcing of a distributed callcenter platform I've been designing/building using FreeSWITCH/Erlang. The goal is to allow multiple callcenter branch offices to operate seamlessly as a whole, or even just scale one large one beyond hardware/software limits by partitioning the inbound lines/agents across multiple servers. You can read more about the design here: http://opencsm.org/wiki/index.php/Spice_Telephony And a quick overview of planned features: * Inbound/Outbound support * Media type agnostic (voice, email, chat, video(?)) * Skill based routing * Agent phone agnostic (SIP or POTS line) * Dynamic wrapup time * Web or fat client interface And you can download a very early work in progress snapshot of the code from http://opencsm.org . We're also open sourcing our customer service management application too, which is also available for download there. If anyone is interested in callcenter development using FreeSWITCH and/or Erlang, we welcome participation at any level. We don't have public source control or a bug tracker yet, but please bear with us; they're on the way. Thanks, Andrew Thompson - opencsm.org

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new ilbc lib
by Alex Gusak 24 Feb '09

24 Feb '09

Hello. After upgrade to version 1.0.3 we have a problem with the codec iLBC (I think that this is due to the transition to a new ilbc libs 1 week ago). Very poor quality for calls to the codec iLBC mode=20 (crack in the dynamic). iLBC mode=30 works well. Tested with phones and Zoiper SJPhone. After a rollback to the old version of FreeSWITCH 1.0.2 this is not a problem, iLBC works fine in both modes (mode = 20 and mode = 30). What could be the problem? -- Alex Gusak

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mod_portaudio: Do not accept next call after Hangup
by Rene Pankratz 24 Feb '09

24 Feb '09

Hello, when hanging up a call with portaudio automatically the next call that is incoming or held is accepted. Is it possible to configure PA that way, that after hanging up (doesn't matter whether caller or callee) no call is activated automatically? I want to choose if I accept the next call or not. Thanks in advance René

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mod_fax and sending a fax
by rod 24 Feb '09

24 Feb '09

Hi all, I don't understand how to use the fax commands for sending a fax. In the wiki I saw this: <extension name="test_txfax_stream"> <condition field="destination_number" expression="^\*90012$"> <action application="txfax" data="txfax.tiff"/> <action application="hangup"/> </condition> </extension> my question is how to specify the gateway/profile that will handle the call. For a call I can use the bridge application like this, but for the txfax ?? <action application="bridge" data="sofia/external/${destination_number}(a)10.10.10.10"/> regards, rod

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mod_openzap stops working after some calls
by Helmut Kuper 24 Feb '09

24 Feb '09

Hello, since yesterday I do a real life test with FS in a 40 sip extension environment with TDM connection via wanpipe/sangoma A104d. I detected two times the problem, that openzap stops working, while SIP calls worked. Only restarting helped. Maybe reloading of mod_openzap helps as well, but I didn't tested that, yet. I filtered FS logfile for hints of that problem and I found a growing number of this line: 2009-02-06 11:09:51 [INFO] ozmod_isdn.c:706 zap_isdn_931_34() Duplicate SETUP message(?) for Channel 1:21 ~ 1:21 in state DOWN [ignoring] Befor restart I saw such a line for *each* TDM channel and no one was able to dial out. mod_openzap handled around 40 outgoing calls until last restart . Since last restart FS runs good for 2 hours now without any duplicate SETUP in log ... Has anybody similar problems? regards Helmut

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