sipX Solution Summary
Posted: May 13, 2005 sipX – Easy to use, scalable, and feature rich open source unified communications and IP PBX solution for Linux SIPfoundry info: Original Website: http://www.sipfoundry.org Github: https://github.com/SIPfoundry sipXcom info: Website: http://www.sipxcom.org sipXcom Wiki: http://wiki.sipxcom.org Github: https://github.com/sipXcom sipX is a…
Read More »Asterisk cmd Directory
Posted: May 12, 2005 Asterisk Directory Command Synopsis Provide directory of voicemail extensions Syntax Directory(vm-context[|dial-context[|options]]) – 1.4 Directory(vm-context[,dial-context[,options]]) – 1.6 Presents the user with a directory of extensions from which they may select by name. The list of names and extensions is…
Read More »Asterisk bounty SMDI
Posted: May 12, 2005 Contributions John Bittner (Simlab.net): USD 3000 looking to use Asterisk to replace voicemail servers used in legacy systems like Comdial. Serial intergration is required with 2 way communication. Email me at [email protected] with any other questions. W. Kevin…
Read More »Asterisk sip tos
Posted: May 11, 2005 SIP.conf: [GENERAL] section: tos= Asterisk 1.2 Set to value or lowdelay throughput reliability mincost none Alternatively you can directly set the bit values: tos=0x18 ; where the tos bits are set individually and in combination For example tos=0x10…
Read More »Asterisk cli srx show layers
Posted: May 11, 2005 Usage srx show layers Synopsis This command shows information about the Sirrix ISDN stack. Example MASTER: port 0x0000l1: port=0x0000, type=BA, mode=TE, ptp=1, ma=1, act=1:l2=08115618, tei= 0, st=’ST_L2_70_MULTIPLE_FRAME_ESTABLISHED_NORMAL’:l3h: cr_out={ }, cr_in={ }l3=0818fef0, cr=( -1, -1), occ=0, ces_m=00000000, hr_m=00000000, hr_s=00000000,…
Read More »TelAppliant
Posted: May 11, 2005 Telappliant / VoIPtalk Official UK Distributor for Digium, Grandstream, IPDialog, Swissvoice Channel Partner for Eicon Networks, Jasomi, Mediatrix, Vegastream, Sipura, SNOM, Speedtouch Internet Telephony Solutions Provider (IP PBX, Bespoke solutions) VoIPtalk Connect – VoIP Termination Service, featuring incoming…
Read More »Asterisk cmd SayNumber
Posted: May 11, 2005 Synopsis Says a number (e.g. “one thousand, two hundred and eighty eight”). Description SayNumber(number, gender) Says number, in the language choosen (defaults to English). Copes only with integers between 0 and 99,999,999. Gender is one of “f” female…
Read More »SER module PA and Internet 2
Posted: May 11, 2005 Internet2 is a consortium of university, corporate, and affiliate members working together to develop and deploy advanced network applications and technologies for research and higher education. In addition to a VoIP working group, it also hosts a Presence…
Read More »SIP tools
Posted: May 10, 2005 Various SIP tools Sip debugging and testing SIPsak: The swiss army knife PlaySIP: Recording a SIP media stream VoIP Recorder: Monitor and Recording SIP media streams. http://www.pcbest.net SIP Server tools Load balancer: SIP Server load balancer SIP development…
Read More »Asterisk settings NeoPhoneX
Posted: May 10, 2005 The following configuration settings worked fine for me: sip.conf [NeoPhoneX]type=peersecret=your-secretauth=md5username=7xxxxxhost=sip.neophonex.hufromuser=7xxxxxfromdomain=sip.neophonex.hunat=noinsecure=verycontext=sip-incomingcallerid=7xxxxxdisallow=allallow=ulawallow=alawallow=ilbc For some strange reasons, NeoPhoneX’s SIP Express Router dials 10069s as your dialin number, instead of s. extensions.conf [globals]OUT_SIP=SIP/NeoPhoneX [sip-incoming]exten => 10069s,1,Macro(stdexten,XX,SIP/XX)exten => 10069s,2,Hangup [sip-pstn]exten => _06xxxxxxx.,1,Dial(${OUT_SIP}/${EXTEN})exten =>…
Read More »Asterisk cmd Ringing
Posted: May 10, 2005 Indicate ringing tone Description Ringing() Request that the channel indicate ringing tone to the user. On an unanswered SIP channel this will send a “180 Ringing” to the endpoint. Example On an answered channel Ringing returns immediately and…
Read More »snom mass deployment
Posted: May 10, 2005 To deploy SNOM phones with an asterisk server try the new res_phoneprov that uses the users.conf file to present all the settings to the SNOM phones. By default the res_phoneprov does polycom phones but is easy to setup…
Read More »Asterisk config zapata.conf.sample
Posted: May 10, 2005 ; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; ;language=en ; ; Default context ; context=default ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100:…
Read More »Asterisk Modules
Posted: May 10, 2005 These are modules you can load or unload in the Asterisk config modules.conf configuration file. In addition to these modules, most applications are also modules you can load or unload. chan_agent.so: Used for the ACD see Asterisk Agents….
Read More »NbooM
Posted: May 9, 2005 ********************************************************************************************* NbooM is looking for buyers to sell its Asterisk-based VOIP solutions at a very reasonable price. This is a golden opportunity to hit the ground running with a VOIP business that has tremendous potential. NbooM’s Asterisk-based VOIP…
Read More »
Phone System 