Hi - We currently have an extremely crappy VOIP provider and want to bring the system in-house. We have a two man tech team (one programmer, one system guy). We have 20 employees across 4 offices.
Here is what I think i need. I hope that you can fill in the blanks:
- One main linux/apache/mysql/php server in a colo facility, running Asterisk (or similar)
- a trixbox server at each branch office (is this necessary?)
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We have an extensive investment in home telehealth equipment with a large patient base using pots videophones and store forward devices and need to accommodate patients with bundled VoIP service. Desirable connection speed with pots is 21kbps with v.34 modems. — Does anyone know of a method / adapter similar to current VoIP ATA's that could be used to support an endpoint with VoIP? — Can the modems be updated to T-38 without sacrificing transmission speed?
I am new to yate . I have registered account in yate client .i have created two extension and made a call .But i cant able to record the conversation between the users. what is the procedure for recording in the yate clients . so any body help me regarding with the process.
Hello, has anyone set up a simple asterisk system with 2 asterisks servers on 2 different subnets? My goal is to call an extension (2000) on 192.168.0.1 from extension (3000) in subnet 192.168.1.1. I would like to use sip. all I want to accomplish is calling between subnets in both directions. I do not have nat nor do I require an ISP. I am on an internal network with subnets. Anyone have sample sip.conf and extensions.conf files? Really simple ones, or files I can use parts that are needed.
I would like these to work:
(sip.conf on serverb;servera would have swapped a to b)
servera type=peer
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