October 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

(no subject)
by Phil Mickelson 03 Oct '13

03 Oct '13

Mick, Thank you for your list. I've search those also. Let me give you a quick example where I actually found the option but the docs don't supply any additional information. I believe I need to create new VM profiles since I'm running multi-tenant. I figured that one out by looking at other commands that expect a voicemail_profile value. In searching through the default profile I found the parameter "callback-context." Since "context" is generally used in FS in connection with multi-tenant I assume (perhaps incorrectly) that this is important for me. So, I go in search of the parameter in the docs. And, after much searching (I have gotten better at this over time) I find: callback-contextdefault <param name="callback-context" value="default"/> That's it. The exact same line in the XML file. Doesn't explain anything. And, if you look at most of those parameters you'll see the exact same thing. Regards, Phil Mickelson Hi Phil, In addition to the books you mentioned, do the following wiki pages not provide everything you need (if they don't we are in trouble! ). API Commands https://wiki.freeswitch.org/wiki/Mod_commands Dialplan Applications https://wiki.freeswitch.org/wiki/Mod_dptools Channel Variables https://wiki.freeswitch.org/wiki/Channel_Variables XML Dialplan https://wiki.freeswitch.org/wiki/Dialplan_XML Documentation https://wiki.freeswitch.org/wiki/Documentation Rgds, Mick Tel/SMS. +44(0)7967 594432

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vars.xml
by Richard Mace 03 Oct '13

03 Oct '13

Hi All, I see that in vars.xml there is an entry for Default Country. Where can I find out what each of the available Default Country options are please? Thanks Richard

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best way to change configuration files programmatically
by crazygabry 03 Oct '13

03 Oct '13

Hello everyone, i'm doing a project and i need to edit xml (dialplan, users and so on) from an external software placed on another server. I saw that with mod_curl can read the dialplan or other configurations from remote places but i don't want to introduce latency when freeswitch parses the dialplan. I came from asterisk and it was possible to change files with Event API, i saw that with mod_event_socket (i think) it isn't possible to change conf files. Can you suggest me the best way to change xml files from remote server? Thanks in advice -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configurat… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch+Radius+IAS/NPS
by Elton Machado 02 Oct '13

02 Oct '13

Does anyone have or know how to setup freeswitch radius module to work against NPS or IAS? I'm try to use the documentation for the freeradius but I'm not succeeding in doing it well. TIA,

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FAX nightmare continues
by Karl Schmidt 02 Oct '13

02 Oct '13

This was tested and worked, but fails in the real world. I have a landline that brings in FAX to a grandstream gxw4104 - which gets connected to a gxw4008 - to a modem to receive faxes via hylafax. There must be too much distortion as I'm getting lots of errors in hylafax and it finally drops the call. I have PCMU on both of the grandstream boxes so I'm thinking that freeswitch should just pass the data. Are there any other settings I might fiddle with? ,.,.,.,.,.,.,.,.,.,. This also brings up a question about echo suppression - there are places in the grandstreams for echo suppression (also in the polycom phones). Should both ends do echo cancellation or just one end? What is the best practice for setting echo cancellation? -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Ask any politician in private, It's a blast spending other peoples money. --------------------------------------------------------------------------------

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OS X works with tarball, not with 1.2 stable git?
by Lawrence Conroy 02 Oct '13

02 Oct '13

Hi Folks, has anyone managed to get the latest git 1.2 stable branch to run correctly on OS X 10.6.8? I have no problems getting the 1.2.13 tarball to build and run, but ... With the 1.2 stable git (as of today), whilst fs builds OK, when run sofia seems to accept regs from clients, will accept an INVITE from the client and send back the 100, set up a worker for the call, but then ignore it. Result is the client sits spinning its wheels waiting for a final response, and getting nothing. Trying to shutdown fs meets a wait for the worker thread to quit, and eventually a time out when the parent gives up. Like I said, this works fine with the tarball. Is there a tagged git release equivalent to the tarball? Without one it's a challenge to track back to find where the problem raised its head. [yup, I am a tyro at this new-fangled git stuff; this may be obvious, but not to me §] all the best, Lawrence

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Documentation - fax_ident and fax_header
by Ali Pey 02 Oct '13

02 Oct '13

Hello, I noticed that the wiki is not correct when it comes to fax_ident and fax_header: http://wiki.freeswitch.org/wiki/Mod_spandsp#By_mod_spandsp_fax.c - fax_ident - fax identity (numeric only) shown on remote machine during communication. Also may be printed on each page. - fax_header - fax identity (alpha-numeric) show on remote machine during communication. Also may be printed on each page. Fax_ident is the Caller's Station ID (CSID) that can be alpha numeric as well. Fax_header would be used to add a header on top of each page on send. I would be more than happy to fix the wiki page, but not sure what the protocol is. I did try to register to be able to edit the page but never got a password. I found the proper explanation on the mailing list from Steve Underwood: > fax_ident should be set to the telephone number to be used within the > fax exchange. This will typically appear on an LCD display at the far > end. In theory it should be limited to digits, spaces, + and one or two > other characters appropriate to telephone numbers. In practice FAX > machines are usually happy with any text, up to 20 characters. This > string may also play a part in page headers. > > If fax_header is set to a non-null string, a header line will be > inserted at the start of each page, just like a typical FAX machine > does. The fax_ident, fax_header and the page number will be used to form > the text of this line. If you are forwarding FAXes, you probably don't > want to add a header line, as there will already be one that was > inserted by the original source. If you are sending a locally generated > FAX, you probably do want to add header lines to each page. Regards, Ali Pey

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ClueCon Weekly - Oct 2 Tomorrow!
by Ken Rice 01 Oct '13

01 Oct '13

Hey Guys, ClueCon Weekly for Oct 2 is rapidly upon us. This week, mod_httapi! We¹ve talked about this before, but this week, I¹ll be demoing how simple it really is to create an application using something as simple as mod_httapi and PHP! Join us tomorrow at 1PM Eastern (that¹s 10AM Pacific) to check out mod_httapi and how easy you can have your WebDevs start creating applications! Where? sip:888@conference.freeswitch.org for more info see https://wiki.freeswitch.org/wiki/FS_weekly_2013_10_02 or just for the calling instructions see: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Using G729 licenses purchased from Digium on FreeSWITCH
by James Mortensen 01 Oct '13

01 Oct '13

Hello, I followed the instructions here http://wiki.freeswitch.org/wiki/Mod_com_g729 to install the mod_com_g729 module. While these instructions which "*explains the installation **quite perfectly" *might work in the general, vanilla cases where things are exactly the same as the documentation author's situation, I suspect mine is slightly different since I'm moving from Asterisk with existing licenses. I've purchased 40 G729 licenses from Digium, but since WebRTC works a lot better on FreeSWITCH, I'm ditching that Asterisk server in favor of FreeSWITCH. I followed the instructions and paste what I believe might be the "sales code" or "activation code" into the validator. It produces an empty licenses.zip file. In the fs_cli, I see "Unrecognised resource G.729A/0", which according to other mailing list threads mean that the licenses aren't valid. The code I got from Digium looks something like this: Key : G729-3543AV54354397WE (not the real code, just an example of what one might look like based on the pattern I see) If anyone has experience porting G729 licenses from Asterisk to FreeSWITCH, please let me know. Thank you! James - *"Being nice to people, building those relationships early on in your career actually helps in making people want to work with you and believing in your vision." - Girish Mathrubootham, CEO/Founder of Freshdesk*

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Re: [Freeswitch-users] Documentation about available options
by Phil Mickelson 01 Oct '13

01 Oct '13

Avi, I appreciate the response. I have found that before but unfortunately, it doesn't answer my question. I'm still searching for complete documentation. Let me add a couple of things also: 1. I have bought all the books: 1.06, 1.2 and the Cookbook. I read the 1.06 book cover to cover. Never did that before and I've been in this business for 35 years. 2. I paid for and attended ClueCon in August. Had a great time. One of the best conferences I've ever attended. I understand this is open source. No, I haven't paid anything directly for the software. I don't expect anything for free. But, if I have no way of getting answers on my own I'm stuck with two options: Either keep searching in the dark, or go back to Asterisk. Asterisk isn't as fast or capable (at least from what I've seen) as FreeSWITCH. One of my big hot-buttons is multi-tenant. After much struggling I've finally be able to get that to work, again, through no help of the docs. I plan to add my own chapter with a full explanation of how to make this work (if I stick around that long!). Sometimes it doesn't matter how good the product is. Thank you again for taking the time to respond. Regards, Phil Mickelson You can send calls to the VM of a specific account, e.g. 1234: <action application="voicemail" data="default ${domain_name} 1234"/> You have to make sure they can check appropriately, too. -Avi On Tue, Oct 1, 2013 at 5:56 PM, Phil Mickelson <phil at cappgen.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: >* Hello,*>**>* Is there anywhere in the documentation where I can go and get a list of*>* options available for any particular section of FreeSWITCH. To be more*>* specific; I was looking for the options that I could use in the Directory*>* XML files. For example, I was looking to find if I could specify the VM*>* box for an extension so that multiple extensions would feed the same*>* mailbox. After much searching I did find the "mailbox" setting. However,*>* this took a long time and I just happened on it in a sample.*>**>* I've now done this for several months and just want to find out if I'm*>* missing something (easy, hard, or I'm just stupid) or it just doesn't exist*>* and I need to continue searching like I do.*>**>* Thank you.*>**>* Phil Mickelson* * *

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