System menu that permits Pana users to access Asterisk features
Posted: October 18, 2004 The system menu * From extensions.conf ; ; A Panasonic handset user can get access to the system menu by ; picking up CO3 or CO4 and dialing Operator (zero). This is provided ; in the contexts co3out…
Read More »Macro to locate a user at up to two external numbers
Posted: October 18, 2004 The FindMe macro ;******************************************************************** [macro-findme] ;******************************************************************** ; ; Dials the localextention then as many as ; 2 remote numbers to find a subscriber. ; ; ARG1 = local estension ; ARG2 = remote number 1 to try…
Read More »Macro to announce inbound call to Pana user or page all
Posted: October 18, 2004 Call routed from Asterisk to Panasonic Extension * From extensions.conf [analog-ext] ; ; This context provides access to internal Panasonicextensions ; ; Panasonic analogextensions ; ; Varialbes assigned are ; PANAEXTxx ………the extension number to dial ;…
Read More »Force VoicePulse or BroadVoice outbound if user selects CO3 or CO4
Posted: October 18, 2004 Pana user selects CO3 on handset From zapata.conf ; provide dial tone to Panasonic CO3 ; immediate=no usecallerid=no usedistinctiveringdetection=no signalling=fxo_ks group=1 ; context=co3out channel=1 ; ring this one for VoicePulse ; From extensions.conf [co3out] ; ;…
Read More »Ring CO3 or CO4 on inbound VoicePulse or BroadVoice call
Posted: October 18, 2004 Ring CO3 on Pansonic KSU for VoicePulse From iax.conf [voicepulse] context=voicepulse-in username=loginname secret=loginpasswd auth=md5 type=friend disallow=all allow=gsm allow=ulaw allow=alaw allow=ilbc host=gw5.voicepulse.com nat=yes From extensions.conf [voicepulse-in] ; ; ; ; This routine handles inbound calls on our…
Read More »Asterisk Compile
Posted: October 17, 2004 Compile Asterisk Ok, you have the source code to Asterisk sitting in your /usr/src/asterisk directory. Now what? cd /usr/src/asterisk make clean make make install Note: By default, Asterisk runs as the root user. This is a security liability….
Read More »Asterisk channel variable DIALSTATUS
Posted: October 17, 2004 Contains a text string signifying result of the last Asterisk cmd Dial attempt: ANSWER: Call is answered. A successful dial. The caller reached the callee. BUSY: Busy signal. The dial command reached its number but the number is…
Read More »myJabber
Posted: October 16, 2004 Product discontinued? Website content missing as of 2009-02-24 See: http://www.myjabber.net The myJabberAE SIP soft phone integrates well with the myJabber IM. SLTS Communications myJabber License Agreement IMPORTANT NOTICE: The free download version of myJabber is intended for use…
Read More »IEEE
Posted: October 16, 2004 IEEE (Institute of Electrical and Electronics Engineers) http://www.ieee.org/ From the IEEE website: Through its technical publishing, conferences and consensus-based standards activities, the IEEE produces 30 percent of the world’s published literature in electrical engineering, computers and control technology,…
Read More »nufone.de
Posted: October 15, 2004 *nufone (This is unrelated to Nufone.) VOIP to PSTN as well as PSTN to VOIP service provider with voicemail. Subscribers to this service get a public phone number that people can call (only in “selected cities” in Germany…
Read More »Intercepting SIP Calls
Posted: October 15, 2004 Intercepting SIP Calls The following discussion applies solely to VoIP using the SIP protocol. H.323 and MGCP are widely used, but they present different isuues when considering call interception. Many ITSP’s are being confronted with the requirement that…
Read More »Asterisk E164 Call Routing
Posted: October 14, 2004 How to Store Asterisk Call Routing Information in the DNS One of the limitations of Asterisk is the dialplan definition. While using wildcards such as NXX can be reasonably flexible, for a flexible dialplan, where you might want…
Read More »SIP method refer
Posted: October 13, 2004 The SIP REFER method is described in RFC 3515 which says in part: This document defines the REFER method. This Session Initiation Protocol (SIP) extension requests that the recipient REFER to a resource provided in the request. It…
Read More »Vovida
Posted: October 12, 2004 Vovida.Org – Open Source Communications Vovida is a communications community site dedicated to providing a forum for open source software used in datacom and telecom environments. Products offered by Vovida include: VOCAL a softswitch A SIP load balancing…
Read More »MVP810 with Asterisk
Posted: October 10, 2004 Making MVP210/410/810 and Asterisk working together. There’s a few parts envolved: 1. Initial configuration: sadly this has to be done with Multitech’s windows only program. Connects the modem cable to MVP810 and an available pc serial port, start…
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