First, download and install Asterisk from www.asterisk.org. To build/install, I used these commands (see their README): make make install make samples Then added these lines to /etc/asterisk/sip.conf: ; Grandstream [1001] type=friend host=dynamic ; cisco phone [1002] type=friend host=dynamic Then add these lines to /etc/asterisk/extensions.conf: exten => 1001,1,Dial(SIP/1001,200,tr) exten => 1002,1,Dial(SIP/1002,200,tr) My phones register as phone numbers 1001 and 1002.

CosmoPBX is bundled with Latest Asterisk 1.4 and Asterisk-GUI beta. You can configure asterisk from web based interface of Asterisk-GUI from web browser. Use following urls to access Asterisk GUI configuration tool. http://[YOUR IP HERE]:8088/static/config/cfgbasic.html (For basic configuration) http://[YOUR IP HERE]:8088/static/config/cfgadvanced.html (For advanced configuration) See similar questions...

Partysip has been tested at an early development stage and for very basic scenario at the SIPit #10 in Cannes (Apr 2002). The list of tested SIP phones is quite long, but partysip has grown since the SIPit #10. The new features has been tested with most common used phones. The regularly tested phones are the Siemens phones, the Cisco phones, linphone (www.linphone.org) and Windows Messenger (be sure to use Windows Messenger4.7! 5.0 don't have support for SIP). See similar questions...

Use the protocol you are the most comfortable with. We prefer SIP. The rule of thumb is - if your server is on a public IP address, not behind NAT router, then SIP is a better choice, Set "nat=no, canreinvite=nonat" in VOIP account settings to enable SIP reinvite to get the best sound quality, if your server is located on a private LAN, then IAX may work better for you, it is more firewall-proof. See similar questions...

No you do not. However, we will be unable to support any other if you have difficulties in using the service. See similar questions...

There are two facilities that can be used to force phone display in the autoVoIP?Troubleshooting and Monitoring Reports:- Load saved phones from an earlier session - this ensures that you do not need to wait for the Phone to re-register to use autoVoIP facilities. Manually add the SIP Server IP address - this ensures that you can monitor traffic to/from the phone in situations where no registrations are being monitored. See similar questions...

Any SIP compliant (rfc 3261) device should work fine. They must support Outbound Proxy. Refer to NATPass™ web site for a list of Interop tested manufacturers. See similar questions...

We have been active participants in the SIPit events and have done extensive tests successfully with a number of 3rd party SIP products directly or indirectly through our customers or partners. A partial list of other products with which we have successfully tested basic and in some cases advanced features include: See similar questions...

You have to configure only one additional field in you Sip device with your Natpass' IP address or FQDN and disable other NAT settings. Most vendors call it Outbound Proxy. For sample screenshots for configuring various devices CLICK HERE : Sipura ; Snom ; Linksys ; Hitachi wi-fi ; Grandstream ; Polycom. See similar questions...

Afelio uses external protocol stacks. Initially, we will be using OPAL for SIP. At a later stage we plan to also have a Sofia SIP based media server. Both stacks are vastly superior to Asterisk's SIP stack and the problems most people experience with Asterisk will simply not materialise. See similar questions...

In the same way that Linux is an open-source computer operating system, Asterisk is an open-source IP PBX. Like Linux, Asterisk has an enthusiastic developer community committed to improving it. It is available on a large number of platforms, including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris. Asterisk is free, can interoperate between VoIP and traditional PSTN systems, and supports the protocols these two systems use, including H. See similar questions...

You configure each section (numbered 1-10) by pointing DSpy to the proper directory on your network. By putting a shortcut to DSpy in your startup folder, it will start with your TSC client during every boot/login and automatically start tracking your TSC progress. See similar questions...

SIP is the Session Initiation Protocol, defined in RFC 3261. It is the most widely used standard for VoIP signalling. It was designed to be a general-purpose way to set up real-time multimedia sessions between groups of participants. SIP is used during and after the call setup. See similar questions...