Free SIP Softphone (Software Phone) If your company is looking to deploy Free SIP Softphones instead of the normal, desktop-based IP hard phone you may be at the right place. The T-MaxSIP softphone technology is compatible with almost any SIP-based IP PBX platforms such as Blue Box, Free PBX, Asterisk, and Trixbox. This Free SIP Softphone can be used on the go with a windows laptop or some windows tablets. Download our most popular SIP softphone, which is completely free to use. T-MaxSIP – desktop portable SIP softphone for Windows OS. The Free SIP Softphone allows you to do high-quality VoIP calls (person-to-person or on a standard telephone) through open SIP protocol. It can also go from the cloud of SIP service provider you can pick what will work for you. Our sip phone is a great replacement for Zoiper, X-Lite (Xlite), Eyebeam, Counterpath and unlike those Free software phones does not come with spyware or ads.
Please Try our Free Auto Dialer Trails
tiny desktop footprint(>2.5MB) and RAM usage (>5MB) – written in C++ and C with minimized system resources used
compatibilityrobustly conforms to SIP protocols
portabilitystores most settings in the ini file
easy to useuser friendly interface in all usage
privacyconfigurable encryption TLS / SRTP for control and media
functionalityvoice, on hold, transfer and dialing
I launch T-MaxSIP but nothing happens.
Check for T-MaxSIP icon in system tray.
How to setup account?
Right click on T-MaxSIP icon in system tray (near clock:).
How to add contact?
Right click on blank white area in Conacts tab.
I cannot connect to server (process registration).
Check SIP server field. Try to change protocol UDP/TCP/TLS
How to specify different SIP port?
If you use SIP proxy – append “:port” to proxy only. If not, append “:port” to “SIP server” AND “Domain”. Format: “proxy:port” OR (“server:port” AND “domain:port”).
I cannot make outgoing call.
Make sure you have entered correct SIP proxy. Try without STUN server. Disable unused Audio Codecs / ICE / Media encryption, why? This feature increases UDP packet size (SDP message length of INVITE query). If UDP packet size will be > 1500 bytes (MTU), it will be fragmented. Not all routers can correctly work with fragmented UDP packets. So, if you enable extra feature like SRTP, or ICE, or select too many enabled codecs, or make video call, be ready that you will not be able make a call. Best exits from situation – use TCP or TLS transport, but in this case your SIP server must support it.
Im getting error while making call: Unable to find default audio device
You must have input and output sound device in your system (speakers and microphone).
I can not hear or not hear me, or nobody hears.
Disable STUN server. Check SIP proxy field.
I use T-MaxSIP without registration on SIP server. How to specify address of my SIP gateway?
You can fill “Domain” in account page OR enter number in format <number>@<gateway>.
How to set up T-MaxSIP for point to point without a SIP server between 2 laptops?
Minimum what need to do – install microisp. Now you can make and receive calls. To make call enter number in format: “sip:192.168.1.33” or just “192.168.1.33”, where “192.168.1.33” – IP address of callee. Optionaly you can specify in account your name, transport and encryption mode, leave fields connected with account empty. TLS transport not works in point to point connections.