Guidance if possible

mmmike

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#1
Hello,
I'm new in the VOIP community and if possible I would like to have some guidance.

I have installed a fresh install of Elastix as a VMWare machine using the Elastix 1.6 64 bits ISO. The installation completed successfully and I have managed to login to the administration website.

I want to use Elastix at home first for testing purposes and move it into my office later. For testing I want to use 2 Iphones with Acrobits software to connect to Elastix and make a phone call. and of course to call to the operator and be able to redirect the call to one of the Iphones from the operator.

If its possible to get some guidance what should I do to make it work and what hardware do I need for that or as I heard I can use a Trunk Dump to make this work.

i just checked on Ebay for VOIP equipment, is this what I need if I dont want to use Trunk Dump : LINKSYS PAP2 SIP VOIP NA UNLOCKED PAP 2

Thanks in advance,
Michael.
 

bhallottawa

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#2
Micheal;

I have about 30 iPhones, iPads and iTouches working with my elastix installation using the soft client from Acrobits.

What kind of help are you looking for specifically?

Bob
 

mmmike

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#3
Well, I would like to know what equipment do I need to run it except the Elastix server that is already up and running (If possible with some refferal to it).
Will be lovely if you have the time to explain what are the basics that I need know to get up and running( hope I dont sound too lazy, its just the job doesnt allow the time to research about it at the moment, I'm pretty sure google has the answers).
 

dicko

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#4
Google works, but "Elastix Without Tears" will largely and concisely explain Elastix/FreePBX/Asterisk's methodologies in a more direct way for Elastix.

It will explain the possible trunk connections to the PSTN (Public Service Telephopne Network) SIP and IAX2 (VOIP trunking, no hardware needed) and Analog/TDM ( traditional "land lines" hardware needed ). So yes and no, the PAP2 will deliver dialtone to a traditional telephone, they provide two FXS (Foreign Exchange Station) interfaces. To connect with a "phone line" you will need an FXO (Foreign Exchange Office) interface perhaps a LINKSYS 3102 device, there are many other options of course.

To get a more general grounding in VOIP try http://voip-info.org .

Unfortunately it will take a little commitment to understand either traditional telephony or VOIP's alternate solutions.

Luckily, Palosanto our gracious sponsors provide "paid-for" support (under the Support tab at the top of this page) , if you are "too busy" ;) )


dicko
 

fred0

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#5
Just FYI, with Asterisk 1.6 (in Elastix 2.0) and iOS4 and the latest version of Acrobits Softphone, you can now set up the client to stay registered and receive calls in the background!

To make it work, you need to have SIP over TCP as described here:
http://www.acrobits.cz/tutorials/ios4-a ... softphone/

Asterisk 1.6 now supports SIP over TCP. To enable it you need to add 2 options to your sip config:
tcpenable=yes
transport=upd,tcp

The first option turns on the TCP support. The second tells Asterisk to use UDP as the primary protocol (to support most sip clients) but to allow TCP as a secondary option if requested (which you can set Acrobits to do as described in the link above). Set this way, all my hard phones continue to work over UDP while my Acrobits client uses TCP.

There's 2 ways to add these options:
A good way is to use unembedded freepbx: enable the Asterisk SIP Settings module and add both in the Other SIP Settings section.
Alternately, you can edit the /etc/asterisk/sip_custom.conf file and put both options there. This method requires that you reload sip. I used "amportal restart" to effect the reload.

A word of warning: sip over tcp is still considered experimental so, if you break your server, don't blame me.

From the file /usr/share/doc/asterisk-1.6.2.8/configs/sip.conf.sample:
; Note that the TCP and TLS support for chan_sip is currently considered
; experimental. Since it is new, all of the related configuration options are
; subject to change in any release. If they are changed, the changes will
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
 

liopleurodon

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Aug 6, 2010
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#6
Re: Re:Guidance if possible

dicko escribió:
Google works, but "Elastix Without Tears" will largely and concisely explain Elastix/FreePBX/Asterisk's methodologies in a more direct way for Elastix.

It will explain the possible trunk connections to the PSTN (Public Service Telephopne Network) SIP and IAX2 (VOIP trunking, no hardware needed) and Analog/TDM ( traditional "land lines" hardware needed ). So yes and no, the PAP2 will deliver dialtone to a traditional telephone, they provide two FXS (Foreign Exchange Station) interfaces. To connect with a "phone line" you will need an FXO (Foreign Exchange Office) interface perhaps a LINKSYS 3102 device, there are many other options of course.

To get a more general grounding in VOIP try http://voip-info.org .

Unfortunately it will take a little commitment to understand either traditional telephony or VOIP's alternate solutions.

Luckily, Palosanto our gracious sponsors provide "paid-for" support (under the Support tab at the top of this page) , if you are "too busy" ;) )


dicko
I'm also new to VoIP, but this has been really helpful to get my server started
 

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