2005 April

April 2005

It\’s not easy to merge a VoIP network and the PSTN network. Recently, I\’ve been looking into how Asterisk handles phone numbers, SIP uri:s and Caller ID names. SIP is based on UTF-8, so it\’s very transparent in how it handles names with non-US character sets. Asterisk happily takes a Caller ID Name, the From: name from SIP utf-8 and stuffs it into the Caller ID Name field, which seems to be ASCII. In Asterisk we have not-so-well-defined \”extensions\” that are alphanumeric in a way, but in a lot of cases expects a phone number only. It will strip off \”un-necessary\” characters, like \”()-.\”. Of course, a SIP address being \”firstname.lastname\” will be changed by these routines. In phone numbers, stripping these characters does not change the phone number. In SIP, it changes the address to something that is a different address. Just one example of many that needs to be fixed.We need to come up with a way to handle this properly and always be aware of the character set, character set encoding and be able to convert from one to another without loosing the ability to place calls and return calls.And it does not surprise me a bit that even in the 21st century, I still have to work with character set handling problems, conversions between character sets and teaching us/english programmers about the need to use national characters like our Swedish \”åäö\”.

During the developer\’s conference call yesterday evening,it was decided that we finally should release the much-awaitedAsterisk 2.0 Stable release, also called \”codename AAFJ\”.This relaese is based on the \”hidden\” cvs that has been inoperation for six months by a group of core development membersin the Asterisk.org open source project, under the leadership ofBrian K. East, who will maintain the stable code base forthe 2.0 CVS tree and releases. (more…)