December 2007 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] bug policy
by Raymond Chandler 30 Dec '07

30 Dec '07

It's recently come to my attention that people are posting bugs in various places on the internet (websites, blogs, etc that seem to have no official affiliation with FreeSWITCH). That, in and of itself, doesn't really strike me as all that bazaar or harmful. I am, however, a bit confused by the fact that the bugs reported in some such forums aren't in jira. Why would someone who claims to be a "member" of the community post a bug to a site not affiliated with FreeSWITCH? Is this an accepted/condoned action? Should a person that posts to a public site rather than to jira expect to receive any support (or inclusion in the project) whatsoever? While I don't claim to be a mind-reader and, hence, cannot tell why a person would do these types of things. I personally feel that the persons responsible for such posts are intentionally trying to defame the project. Since that is my interpretation of the ongoing events, I'm not posting any links to such resources as that would likely just help the author(s) in their goal since i believe googlebot regularly crawls the ML archives. In many projects, there is policy for almost everything. There's policy on where to post bugs and what information to include, where to ask for help, even how many spaces and tabs to use when helping write code. So, what is the FreeSWITCH policy? Is there a FreeSWITCH policy? Is it not thought a good idea to implement such a policy for FreeSWITCH (if it doesn't exist as yet)? If there is currently policy regulating such things, where can one read about it? Since I've "joined" the community, I've tried to help the project in any way I can (reporting bugs that I find, submitting what patches I can, documenting, etc), so I find it a bit offensive that there are people in the community that would do things to hurt the project in any way. By "joined", I mean when I decided that this was a worthwhile project and decided to help where I can, we all know that there's not an actual initiation or anything. I'm pretty sure that the lead devs wouldn't be inclined to approve of the afforementioned actions, but what I really want to know is how others feel (i.e. am I silly for being offended?) and what's expected of a "member" of the community. --ilm

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[Freeswitch-dev] How to make Attended Call Transfer as Transferor??
by Irshad Ahamed 28 Dec '07

28 Dec '07

Hi, I have installed Freeswitch successfully on my machine. I was trying to perform an Attended Call transfer to other endpoint in the same domain as a Transferor. But I couldn't find how exactly I should do that. It would be really great if someone helps me out in telling the procedures to follow. I have tried out Call Transfer as Transferee and it works fine. Thanks in Advance... Best Regards, Irshad.

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[Freeswitch-dev] ATTNETION Javascript DTMF Callback Changes ATTENTION
by Brian West 27 Dec '07

27 Dec '07

http://wiki.freeswitch.org/wiki/Javascript_DTMF_Callback_Function The callback has been changed slightly. In addition to the dtmf.digit you get dtmf.duration. Please refer to the Wiki Page for more information. Thanks, Brian West

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[Freeswitch-dev] How to enable Call Waiting on Freeswitch UA???
by Irshad Ahamed 26 Dec '07

26 Dec '07

Hi, I couldn't hear any beeps when there is a second incoming call. I was of the notion that Call Waiting is enabled by default. If not, how should I enable it?? Any pointers buddies..... Thanks in advance... Best Regards, Irshad.

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[Freeswitch-dev] srtp, nat, speed
by Ray 25 Dec '07

25 Dec '07

Hi Folks, Its my first post here and Merry Christmas. I've just spend the last few days installing freeswitch + wikipbx with the latest svn sources (as of a couple of days ago). It is working. The server is an amd64 vserver on a gentoo. Machine is a dual EM64T 3GHZ with 8GB of RAM with a public IP. Twinkle able to call in from behind a nat and hear the echo test. I then got the speex codec working .It required sofiaconfig.py to be modified on wikipbx: "codec-prefs":"speex,speex@16000k,iLBC@30i,GSM,G722,PCMU@20i,PCMA@20i", I am able to call the sip provider (vitelity) and get an 800 number IVR audio. I am unable to get an incoming DID call answered via twinkle. Firstly it takes a long time to decide to call the extension and the provider hangs up, and then I'm not sure the NAT configuration is right. Is there some dns lookup stalling the extension calling? On my Grandstream BT200 I can get it registered but it does not do the echo test at all - for some reason this works on twinkle. I would like to see SRTP/ZRTP working. I've grepped around the configs and googled but all I can find that looks relevant is in: libs/sofia-sip/libsofia-sip-ua/soa/soa.c where there is an srtp_enable check. How is this enabled in the xml configs? Opportunistic zrtp encryption would be great. Should I bother hooking up an openser + mediaproxy solution for seamless NAT? I was hoping that freeswitch was a complete solution for an internet wide pbx where there would be a mix of NAT'ed and normal clients, and secure conferences would be possible. Cheers, Ray. -- http://www.rayservers.com/

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[Freeswitch-dev] localized dialplans
by trixter aka Bret McDanel 21 Dec '07

21 Dec '07

I have redone the localized dialplan stuff, there is even a readme file. The scope of this is limited, its only to provide a replacement dialplan for what users would be used to in a given country. That way the way they used the PSTN telephone provider they can with freeswitch. Currently there is: NANPA (the 18 or so countries that are country code 1) Netherlands Spain - thanks to David Villasmil Venezuela - thanks to David Villasmil If you want to create a new one for your local area the readme gives full instructions on how. The rumor is that this is going to go into trunk soon so getting this filled out (it only takes 2 minutes) would be nice. look for it in SRC/scripts/contrib/trixter/dialplan in an svn repository near you. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast +44 28 9099 6461 US +1 516 687 5200 http://www.trxtel.com the phone company that pays you!

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[Freeswitch-dev] Conference tones
by Brian West 20 Dec '07

20 Dec '07

I need everyone to help pickout a default tone set for various tones outlined in phrase_en.xml for mod_conference. I'm going to be placing an order for more sound files with gmvoices to fill in any gaps so please try to go over voicemail or anything else that might be missing and let me know what we need to order. Thanks, /b

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[Freeswitch-dev] Call for Help
by Brian West 15 Dec '07

15 Dec '07

I need someone to do some work on this one: At the cli type help... verify each item listed has the proper decription and correct args. For the most part they should be correct on args but the description might not be enough for someone to know what it does. Anyone up for this task? I'm still testing and tweaking the default configuration and have a few more things to add to it. /b

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[Freeswitch-dev] Interop list
by Brian West 15 Dec '07

15 Dec '07

http://wiki.freeswitch.org/wiki/Interop_List Please visit that page and add anything you have done any interop testing with.. Please list dates and such if possible. If you have any hardware you wish to donate to on going interop testing please contact me off list. Thanks, Brian West

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[Freeswitch-dev] Note on using sipphone.com with FreeSWITCH
by Brian West 13 Dec '07

13 Dec '07

Attention everyone: I have added some examples for setting up various sip providers into SVN under directory/default/ that you can use as templates to setup your connections to those providers. One note that I must inform you all on is the fact that SIPPhone.com advertises _sip._tcp.proxy01.sipphone.com that maps to uap.sipphone.com port 443 which is 100% invalid and doesn't work. So when you send an invite to sipphone it'll take 30 seconds for TCP to timeout and then fall back to UDP. You have three options here: 1. Email SIPPhone and ask them to fix it. Which I have done and yet to get a response about. 2. Disable TCP and only use UDP in oubound.xml profile that you register from for sipphone. 3. Do not use SIPPhone at all. The bottom line is they shouldn't advertise an SRV record for TCP if they do not officially support it. Thanks, Brian West

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