IAX Phone
Posted: January 22, 2004 This product appears to have vanished, there is no longer any mention of it on the web site. IAX Phone is an IAX2 soft phone. It is currently available as a free demo. Two GA versions should be…
Read More »Telebau Phones
Posted: January 22, 2004 H.323 phones IP Tel 100: H.323 IP Tel 102: H.323 with 1 S0 gateway (for ISDN BRI) Gateways and adaptors IP Gate 104: 2 x ISDN BRI S0 (max. 6) to 1 x LAN gateway (H.323) IP Gate…
Read More »SER example pstn
Posted: January 22, 2004 SER managing a telephony gateway example: ser configured as PSTN gateway guard; PSTN gateway is located at 192.168.0.10 $Id: pstn.cfg,v 1.2 2003/06/03 03:18:12 jiri Exp $ —————— module loading ———————————- loadmodule “modules/sl/sl.so” loadmodule “modules/tm/tm.so” loadmodule “modules/acc/acc.so” loadmodule “modules/rr/rr.so”…
Read More »SER module enum
Posted: January 18, 2004 Enum module implements enum_query function that makes an ENUM query based on the user part of the current request URI. The function assumes that the user part consists of an international phone number of the form +decimal-digits, where…
Read More »VoIP User Group San Antonio Texas USA
Posted: January 15, 2004 San Antonio VoIP User Group We are in the process of forming the Austin VUG. The plan is to meet once a month for about 2 hours. The proposed meeting format is: Breaking news, 15 minutes Presentation by…
Read More »VoIP User Group Houston Texas USA
Posted: January 15, 2004 Houston VoIP User Group We are in the process of forming the Austin VUG. The plan is to meet once a month for about 2 hours. The proposed meeting format is: Breaking news, 15 minutes Presentation by a…
Read More »Broadvox
Posted: January 12, 2004 Founded in 2001, Broadvox is a worldwide leader in providing integrated managed VoIP services to SMB, Enterprise and Carrier customers. It has deployed one of the largest, full-featured global VoIP networks and is trusted by more than 160…
Read More »Asterisk cmd AppendCDRUserField
Posted: January 11, 2004 Note: This function is depricated. It has been replaced by Set(CDR(userfield)=Value). Synopsis AppendCDRUserField(Value) Documentation AppendCDRUserField appends data to the CDR, Call Data Record, user field. CDR records are used for billing. Though it is not mentioned in the…
Read More »DDI
Posted: January 10, 2004 If you have point-to-point (PTP) ISDN then you may only connect one ISDN device – the PBX – to the S0 bus. In this case the phone numbers are called DDI instead of MSN (point-to-multipoint, PTMP). Other known…
Read More »Asterisk t extension
Posted: January 9, 2004 t (lower case t): Timeout extension in context. T (capital T): Timeout on AbsoluteTimeout Asterisk standard extensions Asterisk cmd ResponseTimeout Asterisk cmd AbsoluteTimeout: Setting the maximum call time Asterisk config extensions.conf: The dial plan Asterisk | Tips &…
Read More »Asterisk cmd AbsoluteTimeout
Posted: January 9, 2004 Synopsis: Set absolute maximum time of call Status: Deprecated in 1.2 in favor of TIMEOUT(absolute) function Removed in 1.4 Description: AbsoluteTimeout(seconds) Set the absolute maximum amount of time permitted for a call. A setting of 0 disables the…
Read More »Asterisk Call Manager for Windows
Posted: January 6, 2004 by Alex Argov Tikalnetworks Crystal Manager Ver 3.3.5 CrystalManager CrystalManager is a call manager and a personal monitoring tool for Asterisk PBX. Extensions status (BLF) Extensions Grouping Call Pickup Call Transfer Conference Call Double-Click to Call Instant Messaging…
Read More »SIP response class1
Posted: January 4, 2004 SIP response codes, class 1: Provisional messages These are sent within a SIP dialogue 100 Trying 180 Ringing 181 Call Is Being Forwarded 182 Queued 183 Session Progress SIP | SIP response codes | SIP response class1 |…
Read More »SIP Response class5
Posted: January 4, 2004 SIP responses, class 5: Server failures 500 Server Internal Error 501 Not Implemented: The SIP request method is not implemented here 502 Bad Gateway 503 Service Unavailable 504 Server Time-out 505 Version Not Supported: The server does not…
Read More »SIP Response class4
Posted: January 4, 2004 SIP responses, class 4: Request failures 400 Bad Request 401 Unauthorized: Used only by registrars. Proxys should use proxy authorization 407 402 Payment Required (Reserved for future use) 403 Forbidden 404 Not Found: User not found 405 Method…
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Phone System 