SIP
From Zenitel Wiki
Contents
Session Initiation Protocol
SIP is the call control protocol used for non-proprietary internet telephony. http://en.wikipedia.org/wiki/Session_Initiation_Protocol gives a brief overview of the protocol. http://tools.ietf.org/html/rfc3261 is the standards document. SIP is a human readable text protocol, which makes it a bit easer to debug.
Configuration Guides
SIP Gateways for analog FXO lines
SIP Gateways for ISDN lines
SIP Gateways for E1/T1 lines
Related standards
SIP is a 270 page document which mainly deals with locating the destination "user" when making a call, and provides a framework for negotiating the media setup. Describing the media (audio/video) formats and protocols is left to another protocol: SDP (Session Description Protocol). SDP messages are attached to some of the SIP messages. See http://en.wikipedia.org/wiki/Session_Description_Protocol for a introduction. http://tools.ietf.org/html/rfc4566 is the standards document. By current practice, SIP and SDP is almost always used together.
RTP (Real-time Transport Protocol) is the most usual method for encapsulating audio into IP packets. Further reading here, http://en.wikipedia.org/wiki/Real-time_Transport_Protocol. And the standards document is http://tools.ietf.org/html/rfc3550. In AlphaCom, all voice over IP uses RTP, both in internal connections with proprietary call protocols, as well as for SIP calls. RTP mainly deals with handling the sequencing and timing of packets, not so much about the actual audio or video contents. However, a number of very common audio codes formats are listed and enumerated in http://tools.ietf.org/html/rfc3551. The number behind the "RTP/AVP" often found in SDP messages refer to table 4 in this document. These are called "static payload types". When they are used, it is possible to playback the stream of RTP packets without having any other information. On the other hand, when using "dynamic payload types", the meaning of the RTP packets has to be agreed upon using an other protocol, like SIP/SDP.
Enable SIP INFO in X-Lite 3.0
AlphaCom use SIP INFO type signaling during connection. The X-Lite has DTMF support for SIP INFO messages, in addition to RFC 2833 and inband DTMF. But in order to use SIP INFO you need to disable the other two options.
Follow these steps on the X-Lite: Dial ***7469 (send)
Once you get the advanced options menu up, filter first for DTMF. Find the Force Inband DTMF entry, double click and change the value to 0.
Then filter for 2833. Find the RTP 2833 enabled line, double click and change this value to 0.
Setting both to 0 will result in your sending SIP INFO DTMF.
Now * and # (and door opening) will work from X-Lite.