Asterisk sound files additional
Posted: August 29, 2005 These sounds were recorded by Allison Smith, who does PBX voicemail voices for hire and is the voice of Asterisk. She does a great job! You can order prompts yourself from Allison by visiting here. This library is…
Read More »Asterisk cmd SetCallerPres
Posted: August 29, 2005 SetCallerPres Set CallerID Presentation This application performs the identical operation to CallingPres(), but accepts descriptive text names instead of numeric values. SetCallerPres(presentation): Set Caller*ID presentation on a call to a new value. Always returns 0. Valid presentations are:…
Read More »Asterisk consultants Canada – Quebec
Posted: August 29, 2005 Aheeva Technology inc. Web site: https://www.aheeva.com/ E-mail:[email protected] Phone: (514) 223-2581 Products: Inbound, SoftPhone, Predictive dialer, Full digital recording Atelka Contact Center Solutions Web site: http://www.atelka.com/ E-mail:[email protected] Phone: (514) 448-4905 Products: Virtual Contact Center Communications Télésignal Inc Web site:…
Read More »Asterisk hardware channel bank check
Posted: August 29, 2005 Considerations for buying a channel bank Channel banks are typically employed if you need to connect ~5-16 analog phones since that is the most cost effective solution. So, if you’re looking for a channel bank to interface phones…
Read More »Free World Dialup
Posted: August 28, 2005 As of August 31, 2010, the domain name (freeworlddialup.com) is redirecting to: http://www.siptosip.net/ which is offering a $100 trial of the Nova high definition SIP service. As of 4/25/2010 the domain name (freeworlddialup.com) has not been paid for…
Read More »Asterisk Avoid SIP NAT Traversal
Posted: August 28, 2005 Using Asterisk as a SIP/IAX Gateway The Asterisk open source telephony server can be used as a gateway in order to avoid SIP NAT Traversal. All VoIP devices on the LAN are configured to connect to an Asterisk…
Read More »Welltech Phones
Posted: August 27, 2005 LAN Phone 201 H.323 and SIP with NAT support PPPoE DiffServ supports both IP telephony and a POTS line (PSTN) Support Mute, Redial, Hold, Transfer, Forward, Speed Dial Management from Web, Telnet, Keypad LCD Menu Motorola PPC 860…
Read More »Suncomm Phones
Posted: August 27, 2005 SunComm Suncomm Lanphone 201 Approx $115USD with a order of 50+ units. Approx $104USD with an order of 200+ units. Approx $98 with an order of 1000+ units. These Phones run on a Motorola PPC 860 multimedia processor…
Read More »Suncomm
Posted: August 27, 2005 SunComm I think the primary OEM produce for this item is Welltech Approx $115USD with a min order of 50 units. These Phones run on a Motorola PPC 860 multimedia processor and run on VxWorks Unix They have…
Read More »PERL-ZAPSIPIAX_INCOMMING_OUTGOING-AGI
Posted: August 27, 2005 // Version 1 (c) for ZAP incomming call file name zapincomming.agi by imran ( Pakistan )// features in this agi are handled: // (1) Call Recording enbabled / disabled database driven controle.// (2) Call forwarding enbabled / disabled…
Read More »Asterisk Bounty record call queues with detailed filename
Posted: August 27, 2005 All completed via the use of a an AGI script on hangup that uses a combination of the Asterisk DB and queue_log to get the requried information to rename the file. HINT – The channel which answered the…
Read More »Asterisk Monitoring
Posted: August 27, 2005 Now that you’ve built Asterisk around your entire life it might be a good idea to make sure it stays up! Argus The current version of Argus has builtin support for monitoring SIP and IAX2, as well as…
Read More »Asterisk sip insecure
Posted: August 27, 2005 Insecure (For asterisk 1.0.9 and above) port: ignore the port number where the request came from invite: don’t require authentication of incoming INVITEs port,invite: don’t require initial INVITE to authenticate and ignore the port where the request came…
Read More »Polycom SoundPoint IP 501
Posted: August 26, 2005 Information and Easy Configuration Guide for the Polycom 501 IP Phone and Asterisk There is good information on this site about setting up a Polycom phone to work with Asterisk, but it’s rather spread out and unorganized, so…
Read More »T.38
Posted: August 26, 2005 T.38 is an ITU standard for sending FAX across IP networks in a real-time mode. FAX messages are sent as UDP or TCP/IP packets. The IETF RFC RFC 3362 implements a media type called image/t38 for T.38 faxes….
Read More »
Phone System 