Asterisk indications BE
Posted: July 6, 2005 Asterisk indications for Belgium Insert this into indications.conf and set the Asterisk cmd SetLanguage to be. [be] description = Belgium ringcadance = 1000,3000 dial = 425 busy = 425/500,0/500 ring = 425/1000,0/3000 congestion = 425/167,0/167 callwait = 1400/175,0/175,1400/175,0/3500…
Read More »Voxeo
Posted: July 6, 2005 Voxeo Improves customer service and lowers communications costs by automating and connecting common enterprise phone calls with Interactive Voice Response IVR or Voice over IP (VOIP) solutions. Voxeo’s VOIP and IVR hosting, turnkey platforms, and developer services –…
Read More »PBX Hunt Groups
Posted: July 6, 2005 What is a Hunt Group? In Computer telephony, a Hunt Group refers to a group of extensions that are organized to process specific calls. Upon answering a call, the PBX ACD may transfer the call based upon the…
Read More »Asterisk authenticate using voicemail passwords
Posted: July 6, 2005 I got full of having to tell users that we couldn’t use their voicemail password in otherapplication, so I wrote this simple AGI script that parses voicemail.conf contents andchecks for a match against provided extension and password. It…
Read More »Asterisk Fax to Email Example
Posted: July 6, 2005 Description This is a working example of a fax to email implementation for asterisk Packages used Spandsp (receive faxes) Bash (script) sendEmail (sending emails) – http://caspian.dotconf.net/menu/Software/SendEmail/ Overview The idea is quite simple. I needed a fax server that…
Read More »OpenSER
Posted: July 6, 2005 Page Contents About OpenSER OpenSER Headlines Headlines Archive OpenSER Events 2008 Events Archive OpenSER Capabilities Download Documentation Tutorials Other Resources OpenSER Modules v1.0.0 v1.1.0 v1.2.0 v1.3.0 v1.4.0 Deploying OpenSER Practical Examples Platforms Resources See also About OpenSER Please…
Read More »About OpenSER
Posted: July 6, 2005 Please note that OpenSER no longer exists. It was forked into two projects, Kamailio and OpenSIPS. The two projects are currently very similar, given that they are built on the same code base, however the two projects have…
Read More »Asterisk call forwarding
Posted: July 6, 2005 If you use this setup a phone can dial *21*<number> for immediate redirect or *61*<number> for delayed redirect, and #21# or #61# to cancel the setting. If you need help understanding how the variables used in these examples…
Read More »Asterisk perl library
Posted: July 6, 2005 The Asterisk PERL Library helps you in developing Asterisk AGI applications that support your dialplan Asterisk Manager applications that can help you monitor Asterisk You’ll find the Asterisk perl library here: FreeBSD misc/p5-Asterisk port In the distribution, there…
Read More »Asterisk config musiconhold.conf
Posted: July 5, 2005 MusicOnHold Configuration Asterisk 1.2 In Asterisk 1.2.x and above you no longer need to use the mpg123 player to play mp3 files, you can use the asterisk addon “format_mp3”. mpg123 or something similar is still required for mp3…
Read More »Asterisk Callgroups and Pickupgroups
Posted: July 5, 2005 Call and pickup groups Letting someone else answer a call In the mgcp, SIP, IAX, Skinny and the zapata channels you can define call and pickup groups for phones. Note that call pickup typically only works WITHIN a…
Read More »Asterisk cmd Macro
Posted: July 5, 2005 Synopsis Macro Implementation Description Macro(macroname,arg1,arg2…) Executes a macro using the context ‘macro-<macroname>’, jumping to the ‘s’ extension of that context and executing each step, then returning when the steps end or a call to MacroExit is encountered. The…
Read More »Asterisk H324M
Posted: July 5, 2005 Asterisk as 3G-H.324M ? SIP Gateway Update to this page: A new Wiki page is created to collect compatibility information about app_h324m: Asterisk app_h324m compatibility Update to this page: As with 2008-01-16 (and long before), H324M calls work…
Read More »H.263
Posted: July 5, 2005 H.263 is a video codec designed by the ITU-T as a low-bitrate encoding solution for videoconferencing. It was first designed to be utilized in H.324 based systems (PSTN and other circuit-switched network videoconferencing and videotelephony), but has since…
Read More »Asterisk phone Windows messenger
Posted: July 5, 2005 Windows Messenger and Asterisk To get Windows messenger to work with Asterisk you need to change the realm in chan_sip.c, the realm needs to be the same as the server name. In Asterisk, the realm is “asterisk” by…
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