Amazon AWS EC2 Web Service can be used for Basic Asterisk IP PBX Installation providng Scalability, Reliability, and Mobility
In this article, I will configure a SIP Phone to connect to the Amazon AWS EC2 instance and make both inbound/outbound calls through a SIP Provider.
I have configured the Asterisk to use Anveo as the SIP provider in Part 3 of the series. However, since then, I have added another provider, nexVortex and am using them exclusively at this time. Article on Implementing nexVortex to your Amazon AWS EC2 Asterisk IP PBX was outlined previously and so was Anveo implementation.
For testing purposes, I will configure Grandstream’s GXV3140 Multimedia Phone and Grandstream GXV3175 Video Phone. Video support will also be enabled to test their video conferencing capabilities.
As previously covered, enabling video is a simple process of adding the necessary codecs and one line to the [general] context in sip.conf file. The codec I have decided on was h264 which is supported by both Grandstream Video phones.
In sip.conf , under [general] context, add the following lines to enable video conferencing capability:
[general]
allow=h264
videosupport=yes
On the phone configuration side, you will have to enable the same codec being used by Asterisk.
Phone’s SIP Account is pretty straightforward and involves the following:
- Account Active: YES
- SIP Server: IP Address of the Amazon AWS EC2 Elastic IP Address assigned to your instance
- SIP User ID: Extension created in sip.conf
- Authenticate ID: Same as the SIP User ID
- Authenticate Password: Password for SIP acconts
- Voice Mail User ID: Usually 2999 or whatever extension you dial to access voicemail
- Name: Caller ID that is displayed
- NAT Traversal: auto
Based on my experience, I have concluded that Amazon’s AWS EC2 cloud is more than capable of handling Asterisk calls for any Small Business VoIP implementation.
I would like to know what your thoughts are on this Subject so I look forward to your comments on this site or via Social Media on Facebook and/or Twitter.
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About Express Interconnect: We are a Portland, Oregon based Small Business VoIP Phone Consultant specializing in implementing quality VoIP PBX system based on Asterisk IP PBX. We can be reached at 503-820-3513 or via email at: support@ExpressInterconnect.com and can help with any questions you might have.
Due to Demand, we have finally branched out into Hosted VoIP Phone Service for Small Businesses. Our Hosted VoIP Service provides full Asterisk IP PBX functionality in a simple Plug and Play system.
Also, please make sure to visit our on-line store at VoIP Phones Direct for all of your IP Phone needs. We carry Grandstream, Polycom, Snom, Aastra, and Yealink IP Phones with a 14 day money back guarantee and FREE UPS Ground Shipping.
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