May 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Answering Machine Detection
by David Brazier 30 May '08

30 May '08

Hi All Just wanted to clarify: * Is there any means of answering machine detection in FreeSwitch (along the lines of Asterisk app_amd)? * If not, could something like that be implemented in a custom app (I'm not sure if there is sufficient access to the media stream?), or does it need to be in the core? * If in the core, switch_ivr_record_file seems like a good starting point, as it already does speech detection. I would hope I would be able to implement something starting from switch_ivr_record_file, but I didn't want to charge in at that level if it would be better done elsewhere. Any thoughts appreciated. David PS Great project - a real credit to the open source community! 360CRM Limited, registered address, Caledonia House, 89 Seaward Street, Glasgow, G41 1HJ, Registered in Scotland, company number SC261952. This email and its attachments may be confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of 360CRM Ltd. This email does not form any part of a contract. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error.

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Max of 170 channels in the conference room.
by Johny Kadarisman 30 May '08

30 May '08

Hi all, I build a new test boxes for freeswitch. and trying to simulate high load condition. I have one freeswitch box that host conference apps, and another box with simple js to originate call to the conference room. To monitor the sound quality, I have one phones that dial into the conference room, and then start streaming/playing moh music in the conference. In many scenario that i tried (mute, unmute, one or multiple rooms), the max total number of channels is about 170 channels before FS start to slowing down on creating new channels. The Cpu's only tops around 40-50%, and the voice/rtp start to jitter or delays on my phone. These test boxes is running on 2 processor 2.8GHz Dual Core xeon cpu's and about 1Gb memory. I think the number already pretty impressive, at least for me :) But is this limit sounds right from others experience? Can I still do more tuning? Thanks, Johny K.

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RESTful Bounty ..... Putting the competition to REST with RestFul VoIP2.x services
by EdPimentl 29 May '08

29 May '08

I am putting a bounty to add RESTful WebServices in FreeSwitch. The bounty ask that TONIC is use as the RESTful Framework. TONIC should then be use to create simple CALLback application and establish ad-hoc VoiceConference. To those not familiar with TONIC here is write up. Tonic <http://tonic.sourceforge.net/> A *REST*ful Web App Development Framework Tonic is an open source less is more, RESTful Web application development and Web site management PHP script designed to do things "the right way", where resources are king and the framework gets out of the way and leaves the developer to get on with it. Tonic helps you develop Web applications that embrace the way the Web really works, enabling your applications to scale, extend and work with other systems easily. Everything is a resourceTonic works exclusively with resources, a resource is simply a piece of data with some metadata accessable by a URL using HTTP methods. Everything within a Tonic system is a resource and so can be handled via the standard HTTP interface.Domain modelling through resourcesWhen modelling your problem domain within a Tonic system, every domain object is a resource or a collection of resources.Keep it simple, stupid!Following the Unix philosophy of many small parts working together via a common interface, Tonic uses resources as it's "programs" and HTTP as it's "streams" to enable you to build complexity out of simplicity. Main features Tonic embraces a number of standard Web features and best practices, as well as features, principles and ideas taken from PHP, Unix, and other software design patterns: - Resource editing via HTTP PUT and DELETE methods (or via the Resource::save() and Resource::remove() PHP methods) - Easy to extend via new resource classes - HTTP basic and digest authentication - Resource method permissions to provide authorisation to individual resource methods - Output templating via Smarty - Output caching via HTTP 304 response codes and modified headers - Extendable resource storage types via adapters - Content negotiation - Match multiple URLs to a single resource - Follows Unix modular design principles And here is why everything one does or will do, should be done via RESTFul messaging.. Think about it, Google, AMAZON, YAHOO have standardized on REST. http://www.infoq.com/articles/rest-introduction http://wiki.opengarden.org/REST/REST_for_the_Rest_of_Us http://en.wikipedia.org/wiki/Representational_State_Transfer http://rest.blueoxen.net/cgi-bin/wiki.pl?FrontPage http://www.jibbering.com/2002/4/httprequest.html http://java.sun.com/developer/technicalArticles/WebServices/restful/ http://www.ics.uci.edu/~fielding/pubs/dissertation/software_arch.htm#sec_1_2 ****** -E http://mobiquity.ws

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IVR: Techniques for logging DTMFs to CDR
by Michael Collins 29 May '08

29 May '08

Gang, I'm working on my outbound IVR and I'm wondering if you have any thoughts on this topic: I'd like to make a record of the digits dialed by the called party. For example, I have an IVR menu that says, "to repeat these options, please press star." I'd like to record how many times the called party pressed star, along with all of the other digits. What do you suggest? One idea is to create a custom channel variable, like "digits_recd" and make it a string of digits, appending a new digit to the string with each key press on the part of the called party. I do like this idea because it preserves the order in which the digits are received. Any other suggestions? I'd like to know what you've done. BTW I'm not using any scripting - this is all being done with the XML dialplan, phrase macros and ivr menus. Thanks, Michael

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Re: [Freeswitch-users] RESTful Bounty ..... Putting the competition to REST with RestFul VoIP2.x services
by Ward Mundy 29 May '08

29 May '08

> On Tue, May 27, 2008, Michael Collins wrote: > > > Re: Any chance we'll see some of your sweet recipes on nerdvittles with a little FS baked in? :D > You bet. In fact, we're raising a bounty on the trixbox forums to port FreePBX to FreeSwitch as we speak... http://trixbox.org/forums/trixbox-forums/open-discussion/freeswitch-1-0-0-p… Raised $3,000 in the first afternoon!

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Conditions in the dialplan
by Jonathan K. Creasy 29 May '08

29 May '08

I've been looking around a little bit and I haven't seen anything on this yet so I wanted to ask here. I want to have a dialplan that captures certain SIP responses and takes different actions based on the response received. For example, if I place a call out to a peer and get back "486 Busy here" I want to do "action1". If I get "302 Temporarily Unavailable" I want to execute "action2". What's the best way to go about this? -Jonathan No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 269.24.1/1469 - Release Date: 5/27/2008 1:25 PM

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Re: [Freeswitch-users] SRTP in PhonerLite and Freeswitch
by Brian West 28 May '08

28 May '08

Yes that is correct.. you say 8000 but its really 16000 anyone that says 16000 is doing it wrong. /b On May 28, 2008, at 1:01 PM, Alfred E. Heggestad wrote: > Brian West wrote: >> What else is great is his phone does G722 wideband. w00t! > > btw, what kind of G.722 wideband codec do you support? > > I see that you are using G.722 implementation from the "voipcodecs" > library, and that you are using different samplerates, one signalled > in SDP (8000 Hz) and 16000Hz as the actual DSP samplerate. Is this > correct? > > From RFC 3551: > > 4.5.2 G722 > > G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding > within 64 kbit/s". The G.722 encoder produces a stream of octets, > each of which SHALL be octet-aligned in an RTP packet. The first > bit > transmitted in the G.722 octet, which is the most significant bit of > the higher sub-band sample, SHALL correspond to the most significant > bit of the octet in the RTP packet. > > Even though the actual sampling rate for G.722 audio is 16,000 Hz, > the RTP clock rate for the G722 payload format is 8,000 Hz because > that value was erroneously assigned in RFC 1890 and must remain > unchanged for backward compatibility. The octet rate or sample-pair > rate is 8,000 Hz. > > > ... > > are you planning to add G.722.1 support at some point? I guess there > are no free implementations of this codec, yet.. > > > Thanx > > > /alfred Brian West sip:brian@freeswitch.org

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SRTP in PhonerLite and Freeswitch
by Krzysiek 28 May '08

28 May '08

Hi I have 2 softphones PhonerLite (they support SRTP via SDES ) and the freeswitch (windows RC1 version) server and I wanted to make secure call between those two endpoints (SRTP). I spend whole day on testing this scenario and my conclusions are: - when the option: <action application="export" data="sip_secure_media=true"/> is uncommented, and both enpoints have enabled SRTP then: 1) Initiator of the session sends SIP Invite with a=crypto paramter and supported codecs 2) Freeswitch receives SIP Invite and sends SIP Invite to the receiver (also with the crypto) 3) Receiver receives the SIP Invite with the a=crypto parameter and he sends back supported codecs with 200 OK message (but without a=crypto parametr. Is that ok? I'm afraid not) 4) Freeswitch sends 200 OK message but witout any codecs: m=audio 0 RTP/AVP 19 and no a= parameters! 5) Final result is that the second leg of the session between Freeswitch and receiver has SRTP transport enbaled and the first leg (initiator- Freeswitch) doesn't hear anything - no codecs! However Freeswitch is sending RTP (not SRTP) pacekets to the initiator. Could someone explain to me, what is going on, and why freeswitch doesn't forward codecs accepted by the receiver to the initiator? Is it a PhonerLite's bug or freeswitch's? Maybe someone has tested SRTP with the PhonerLite softphone or any other free softphone with srtp support? When I uncommented: <param name="Inbound-no-media" value="true"> everything works fine. The parameter <action application="export" data="sip_secure_media=true"/> doesn't change anything then (but i cound miss something). Thanks for help Chris

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SS7 support in FS
by Alex Kinch 28 May '08

28 May '08

Hi all, A colleague of mine who currently uses Asterisk was asking me the other day about SS7 support in FreeSWITCH - and whether people have such a thing in production. I don't think he's looking for MAP support, which makes life a little easier. I know of at least one company that's used FS with TelcoBridges hardware to achieve an SS7 interconnect, but thought I'd throw the question out to the list for discussion. Thanks, Alex

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SS7 and SIP
by imthiyaz＠peopletech.co.in 28 May '08

28 May '08

HI We want to try generate 5000 simultanious Voice broadcast calls . can the below config will work? SS7 Links< ----> Sangoma SMG<------->FreeSwitch<-------->Broadcasting Application ( SIP based) Thank you Imthiyaz -------------------------------------------------------------------- mail2web.com What can On Demand Business Solutions do for you? http://link.mail2web.com/Business/SharePoint

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