Author: VoIP Info
IPmenta
Posted: March 1, 2004 Maunfacturer of VOIP gateways and USB handsets. Quote: They use Taiwanese proprietary codec Peer to Peer protocol (not even going through proxy server). All RD are located in Hsin-chu Taiwan (that’s where a lot of Taiwanese IT labs,…
Read More »Asterisk setup medium office 100
Posted: February 27, 2004 Jan. 2004, by mattf Our max for a single machine is 40 concurrent SIP -> Zap conversations for about a 12 hour period and over 5000 total phone calls per day. We didn’t see crashes going over that,…
Read More »Asterisk setup success 5
Posted: February 26, 2004 OACYS Technology 2/25/2004 We are using 12 phones in the office with one spare. All are SNOM 200 model phones (more about this later) that we purchased from ABP Technology (http://www.abptech.com/). We have a total of seven co…
Read More »SIP method subscribe
Posted: February 23, 2004 From RFC 2848: When a SUBSCRIBE request is sent to a PINT Server, it indicates that a user wishes to receive information about the status of a service session. The request identifies the session of interest by including…
Read More »Asterisk wiki
Posted: February 22, 2004 Asterisk wiki Guidelines for authors I’ve tried to standardize the way I created links to make it simple to create crossreferences. All links that are specific to Asterisk start with Asterisk as a keyword, to differ between other…
Read More »Sipsak
Posted: February 22, 2004 sipsak is a small command line tool for developers and administrators of SIP applications. It can be used for some simple tests on SIP applications and devices. Features sending OPTIONS request sending text files (which should contain SIP…
Read More »SIP method info
Posted: February 20, 2004 From RFC 2976 The intent of the INFO method is to allow for the carrying of session related control information that is generated during a session. One example of such session control information is ISUP and ISDN signaling…
Read More »sipc
Posted: February 15, 2004 http://www1.cs.columbia.edu/~xiaotaow/sipc/ SIP softphone from Columbia University Runs on a range of platforms: Windows 95/98/NT/2000/XP, Linux and Solaris. sipc does not provide audio and video functionality itself; rather, it uses external media application for handling media streams. Currently, it…
Read More »PBX Call Waiting
Posted: February 13, 2004 Alerts the user of an incomming call while participating in another call, and allows the user to switch back and forth. This is most common on a single line telephone in residential or small business settings, as a…
Read More »zaptelBRI TE mode zapata.conf
Posted: February 13, 2004 [channels] ; ; ISDN quadBRI interfaces ; switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local group = 1 context=to-pstn channel => 1-2 switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = local group = 2 context=to-pstn channel =>…
Read More »Asterisk setup soho 16
Posted: February 12, 2004 Non-profit European youth NGO With ASTERISK the Brussels-based headoffice aims at drastically reducing the costs for international phone calls, while allowing remote chapter offices and individual members to call in through VoIP (Free World Dialup) – voice communication…
Read More »SDP
Posted: February 12, 2004 SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. SDP is used from VoIP signalling protocols like SIP, H.323 and some minor VoIP protocols to…
Read More »Asterisk “s” extension
Posted: February 11, 2004 The “s” extension is used when there is no known called number in the context used. The “s” extension is used when starting a call. It is also used when defining a macro. Incoming calls are always placed…
Read More »Getting Gnophone to work
Posted: February 8, 2004 GnoPhone (http://www.gnophone.com) is a linux based soft phone. If you run linux it is one of the best ways to play with Asterisk. Some important starting points GnoPhone is a IAX client, not a sip client. Please do…
Read More »Asterisk sip fromuser
Posted: February 8, 2004 Fromuser= and fromdomain= This is used when calling TO this peer FROM asterisk. If you’re using _register=>_ with another SIP proxy, this setting can come in handy since some SIP networks only allow users in the right domain…
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