Author: VoIP Info
AX-400P
Posted: November 15, 2006 4 ports FXO/FXS card AX400P is a four FXO/FXS ports telephony card with PCI-E interface.Using AX400P analog card, open source Asterisk PBX and stand alone PC, users can create their IPPBX telephony solution which includes all the sophisticated…
Read More »trunkCall
Posted: November 15, 2006 A residential VOIP based company with the vision to bring the loved ones close. harnessing many years of operating experience in the international wholesale voice market where the best prices are only available to large companies and telecom…
Read More »Asterisk Consultants Pakistan
Posted: November 15, 2006 Vocal Solutions Complete Asterisk Support in Pakistan and all over the world. Complete Call Center Setup. IP pbx implementation with DID and ACD support. Whole termination billing server implementation and call card tern key setup. DID of 30…
Read More »SoundPoint IP 601 Expansion Module
Posted: November 14, 2006 The Polycom SoundPoint IP 601 Expansion Modules (sidecars) act merely as “overflow” for contacts which won’t fit into the phone’s own contact/line monitoring buttons to the left of the LCD display. In other words, you can’t put contacts…
Read More »Web based Asterisk Database maintenance
Posted: November 13, 2006 I wrote this PHP script in order to manage the asterisk database without using the CLI. It uses the Manager interface and can thus run as any user without sudo. You must edit the PHP script providing it…
Read More »DIDX
Posted: November 12, 2006 DIDx or DIDXchange is the world’s first and largest DID/DDI (Direct Inward Dialing Numbers) exchange. It made its debut with 100 future-thinking ITSP members in August 2005. Just over one year later, it was 3500+ members strong. With…
Read More »IAX2 Trunking Providers
Posted: November 12, 2006 This is a list of IAX2 trunk providers from around the world. Tpad provides high quality IAX2 Trunking services to businesses around the world. http://www.VirtualPhoneLine.com World’s first company to provide a Phone Number on an IP Device in…
Read More »The-Receptionist-Console
Posted: November 12, 2006 The Receptionist Console for Asterisk provides an optimized user interface for Receptionists who have to handle large numbers of calls. Designed for organizations with a dedicated receptionist who is responsible for taking and forwarding large numbers of calls,…
Read More »AUTO PROVISIONING EQUIPMENTS (GXP2000/PAP2)
Posted: November 10, 2006 Program for windoz to provision voip equipments. Provisioning tool to generate files for many phones at the same times (XML, grandstream …) distribution by HTTP or ftp. For Grandstream (GXP2000…)Parameters generationPhone book GenerationReboot For Linksys (PAP2…) (And by…
Read More »CallWeaver
Posted: November 10, 2006 Synopsis CallWeaver (formerly known as OpenPBX.org) is a community-driven vendor-independent cross-platform open source PBX software project. It was originally derived from Asterisk, a Linux software-PBX. CallWeaver is a fully featured PBX in software. It supports analog and digital…
Read More »WorkSpace3D
Posted: November 8, 2006 Unlimited meetings and cooperative work – Voice, video and data 100% over IP: Each participant can send his video, speak and share. – Unlimited organizers: You can have as many meeting organizers as you want. – Several meetings…
Read More »GIX100
Posted: November 8, 2006 EASY3CALL SIP ATA EASY3CALL lists two models of SIP ATA. The GIX100 and GIX101 All our SIP ATA Ethernet ports are 10/100 Mb/s that can be configured as either switched or routed ports, built-in NAT router/DHCP server and…
Read More »GIP300P
Posted: November 8, 2006 EASY3CALL SIP phone EASY3CALL lists three models of SIP phones. The GIP300, GIP300P and GIP300D All our SIP phone Ethernet ports are 10/100 Mb/s that can be configured as either switched or routed ports, built-in NAT router/DHCP server,…
Read More »GIP300
Posted: November 8, 2006 EASY3CALL SIP phone EASY3CALL lists three models of SIP phones. The GIP300, GIP300P and GIP300D All our SIP phone Ethernet ports are 10/100 Mb/s that can be configured as either switched or routed ports, built-in NAT router/DHCP server,…
Read More »Asterisk sip silencedetecthangup
Posted: November 7, 2006 sip.conf Sielence Detection: silencedetecthangup Syntax: silencedetecthangup=X i.e. [jane-7960] type=friend secret=foo context=inbound silencedetecthangup=25 This command is used to set a time out for asterisk to end a call after a certain amount of secconds that it “hears” complete silence…
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