October 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Sending and Receiving faxes - freeswitch
by Ravi 01 Oct '13

01 Oct '13

Hello Everyone, I am trying to send and receive faxes. I have assigned the number 4302050 for receiving faxes, and the use configuration is given below: <include> <user id="1050"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1050"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1050"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1050"/> <variable name="effective_caller_id_number" value="1050"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> The dial plan that I have in place to receive faxes is given below: <extension name="fax_receive"> <condition field="destination_number" expression="^(050$)"> <action application="set" data="ignore_early_media=true"/> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="$${base_dir}/faxes/${caller_id_number}.tif"/> <action application="hangup"/> </condition> </extension> When this fax number is called from another telephone, I hear the fax dialtone. But when the fax number is called from any fax machine and try to send a fax, it gives a number not available message. I do not recieve this call in the freeswitch server as well. Would really appreciate if anyone could help me in this, to let me know if I am missing something. Thanks in advance. Ravi

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Documentation about available options
by Phil Mickelson 01 Oct '13

01 Oct '13

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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RTP Audio Stream Initialise
by Callum Guy 01 Oct '13

01 Oct '13

I am trying to understand how audio streams are initialised in FreeSWITCH. I'm working with ESL (inbound and outbound) and trying to record some audio. Through testing i have proven that audio is working correctly for bridged calls, conferences and file playback however not for single legged audio recording - that is, until I play a file first. In my example the playback application is executed from an outbound socket, the call is then parked and the inbound socket captures the park event and starts recording. With the playback in place i get audio, and without it I simply get a silent file of the correct duration presumably meaning that the the RTP stream is not active. My question therefore is: How can I initialise the RTP stream for recording, without having played a file first? Is there a command for this, or an argument I can pass to the record application? My objective is to understand how RTP streams are initialised in FreeSWITCH and any input would be greatly appreciated.

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lua session() is blocking...
by Antonio Silva 01 Oct '13

01 Oct '13

Hi all, I was trying from a lua script call a remote destination and when it starts ringing i want to bridged it with another destination. The problem is that the freswitch.session is blocking and it only returns when the call is already answer... Is there a way, beside doing it by events, to get the complete status of a session? the lua script: dialstring = "{ignore_early_media=true}loopback/100/outgoing" session = freeswitch.Session(dialstring); -- check if is ringing... uuid = session:getVariable('uuid') state = api:executeString('eval uuid:' .. uuid .. ' ${Channel-Call-State}') mlog.warning(state) while (session:ready() == true) do session:sleep(500) mlog.debug('in session') end

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No ringback for calls from cell phones
by Siri MM 01 Oct '13

01 Oct '13

Hi, I am using FreesWITCH with Sangoma 102 and E1 connection. When I make a call from a fixed number to one of the internal extensions, ringback can be heard on the fixed number. However, when I make a call from a cell phone, although the internal extension rings, I cannot hear any ringing on the cell phone. I have tried two different mobile phone operators, but the result is the same. I have tried specifying the channel variables such as ringback and transfer_ringback int he dialplan, but doesn't help. Any pointers on how to debug this? Could it be the telco, in between the cell phone operators and FS, causing this? Thanks!

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Differentiating internal/external calls
by Siri MM 01 Oct '13

01 Oct '13

Hi, Is it possible for me to have two sets of dialplans - one for incoming (to FS) and other for outgoing (from FS), in the same context? On receiving an incoming call, I want to run through only a subset of xmls, and give up on not hitting, and the same for outbound - is this feasible? Or any channel variable that would help me differentiate between a external call and internal call? Thanks!

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Minimal configuration for new projects
by Stanislav Sinyagin 01 Oct '13

01 Oct '13

I got bored with cleaning up the vanilla configuration one more time, so I created this Github repo as a base for new projects: https://github.com/xlab1/freeswitch_conf_minimal This is a FreeSWITCH configuration which is completely free from features and predefined stuff (with a small exception in the public context). feel free to use as you like.

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CALL-ID
by Miha 01 Oct '13

01 Oct '13

HI, I need a little help. I have opensips and behind opensips a hew few Freeswitch servers which are for media handling. Opensips is also doing acc with radius. Here is what is happening and I do not know how can I prevent FS do doing so: A CALLS B. A(call-id)->Opensips(call-id)->fs(call-id)->opensips(fs change call-id!!!)->B(changed call-id). How to prevert so that FS will not change call-id of the call? thx! miha

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