SIP
Articles in SIP (10)
- SIP with firewall/NAT using AsteriskNetwork Address Translation (NAT) is a common practice used in networks, and it doesn't play well with VoIP. Solving this problem requires an understanding of NAT, VoIP and your VoIP setup. The following focuses on the SIP protocol for VoIP using Asterisk, but problems and solutions are applicable to most other situations.
- How do I Define the CODEC which I Want to Use in DIDWW SIP Trunks?
This article will help you to define the codec you would like to use with your system.
- Is it Possible to Set Two Different Mappings to Different SIP Servers, for Backup Purposes?
- How does my SIP Server can Identify the Calls?
- Can I Use 3rd Party SIP Services such as Callcentric.com, Sipgate.com or Phonegnome.com?
- Is it Possible to Forward an Incoming DID Number to the SIP Server of my VoIP Provider?
- Can you Recommend on a Free SIP Server that Works on Linux?
- Do DIDWW Systems Accept Non-Standard SIP Ports Such as 5070?
- How Do I Configure my SIP Phone to Work with DIDWW?
- How to Set SIP Trunk Configuration for DIDWW DID Number on Asterisk?
The following guide will explain how to set new DID number on Asterisk.
