VoIP-Forum

The Asterisk project has a very strict policy in regards to backwards compatibility. Unless we can’t find another solution, we’re not allowed to remove a function between releases. A configuration for Asterisk 2.4 should work in the next release. In order to be able to change functionality we warn users in one release and then remove the functionality in the coming release. So a configuration in 3.0 works in 3.2 but maybe not in 3.4.

This article tries to provide help with known problems with upgrading. Read on to learn how to avoid the traps! (more…)

For many years, I’ve been working to remove the “type=user” and “type=friend” from chan_sip. In 1.4 you can in fact do everything you can do with a user with a peer. The only difference is the way we match incoming calls.

• For users, we match the user object name with the From: username (without the domain).
• For peers, we match on IP for incoming calls.

I’ve created a branch called “kill_the_user” that has no type=friend or type=user, only peers.alls are handled this way

• First, we match on peer object name with the From username
• Then we try to match on IP/Port
• If we can’t match, we send to the context defined in the “general” section in sip.conf or to “default”.

If you can find any way this may give you problems, please inform me now. Otherwise, I’m goingto test this branch with all of you. I don’t foresee any problems going ahead with this in trunk.s compatibility, we will accept “type=friend” and “type=user” with a warning in 1.6,then remove it totally. Old configurations will still work. The longterm goal is to define new objects, please check the docs for Codename Pineapple objects,phone, service and trunk.For realtime, realtime users will not be used any more.I hope this will make chan_sip easier to understand and use. Feedback is appreciated!
/Olle

In the stream of updated drafts for the IETF meeting in december, I found a draft namedReal-time text interworking between PSTN and IP networks.This is work-in-progress in Asterisk svn trunk. We already have support for T.140 realtime text in the SIP channel, and support for US TDD phones in the Zap/PSTN channel driver. During this summer, I’ve assisted a developer who implemented a RTT gateway in Asterisk, so you could call with an old PSTN TDD phone and have a conversation with a modern SIP client with support for realtime text.This code will soon be implemented in Asterisk svn trunk, so Asterisk will become the reference implementation for this draft. How many xOIP acronyms do we have now? FOIP, VOIP, TOIP? I surely missed a few there… :-)From the abstract:

IP networks can support real-time text communication. SIP-based real- time text is called Text-over-IP or ToIP. PSTN networks support real-time text using textphones (or TTYs). When real-timetext is supported by different networks, gateways are needed to provide interoperability. Real-time text capable gateways may also support real-time voice.This specification describes procedures for interworking between ToIP and PSTN textphones using a real-time text capable gateway (RTT gateway). It also describes ways to route calls to RTT gateways for several call scenarios. Procedures that support the phased introduction of RTT gateways and procedures that support the invocation of text channels at any time during the call are included. Interworking of PSTN text phones thatdo not support simultaneity of voice and text with IP User Agents that support simultaneous voice and text is also described.