Hey everyone, I have been tasked to set up asterisk on a laptop with the intention of being able to place calls over SIP/AIX. Because of the nature of the calls we will be placing we cannot use a third party provider for VOIP services because it is my understanding that most/all providers record calls that are placed using their service and we are dealing with confidential client information. I am extremely new to asterisk but have a strong background in linux.
Our office currently has its own voip system that can be connected to and a gigabit internet connection (I basically can have a IP deskphone that I can use but have no access to the main system or servers). Is it possible to get this laptop set up as a asterisk server in order to place calls over SIP/AIX without the use of a third party VOIP provider?
(I apologise in advance if this post doesnt make any sense, I am very new to this technology)
After doing research all day I want to try and refine my question to make it more clear.
I have a basic asterisk install set up on a laptop that has analog port and Ethernet ports. Our company has some sort of VOIP phone system(I am not internal IT so I don’t know any specifics) with an IP phone at my desk. Is it possible to configure my asterisk server to connect to my office VOIP network and place calls to the outside world. The main reason I am doing this is for the feature in asterisk that allows caller ID spoofing for social engineering assessments I preform.
Hope that clarifies things.
You need to know if your system accept SIP connections, if yes(SIP is the most common protocol for VoIP) you can create “trunk”/peer between your asterisk and your system, then all is dialplan programming.
No-one is going to want you spoofing caller ID, and, if they have any sense, they will configure their systems to prevent it.
ITSPs are only going to record actual speech to the same extent as PSTN operators, basically only if a court tells them to do so. To do otherwise would be a major waste of resources. Both will, of course log traffic flow data, because they need it to support billing.
To me “social engineering” means acting as confidence trickster.
Thanks for David and Navaismo for the info! I will try and set that up.
Does anybody know what protocol is typically used with ShoreTel hardware (i have a shoreTel230 at my desk)?
Shoretel’s native protocol for communications between their gear and their phones is MGCP. Asterisk’s support for MGCP isn’t stellar, and Asterisk can’t act as a client, only as a server.
Shoretel does have some SIP support, but I don’t think that extends to their endpoints, just as a trunking protocol between their switches.
Cheers