Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ??
Where can I find it?
Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All,
Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
2007-06-29 - AsteriskC2D Launched - Click to Dial from Almost any Mac Application!
2007-06-29 - CEO of Cubic Telecom interviewed - Pat Phelan talks about roam4free and Cubics new Wifi VoIP/FMC product due for release later in the year
by dialexia on Wednesday 15 of December, 2004 [15:05:05 UTC]
Dialexia Corporation announced today the release of their new software Dial-Gate with IP-Tone® services.
December 13, 2004
Montreal, Canada
Dialexia Corporation, a leading provider of VoIP Traffic Management and Billing Solutions for the ITSP, announced today the release of their new software Dial-Gate with IP-Tone® services. Dial-Gate® is a Web-based SIP proxy and a centralized routing server for both the VoIP & PSTN networks. Among its many features are; A Pre/Post Paid Billing System, Real Time Monitoring, and CDR Management. With the added option of IP-Tone, Dial-Gate can handle incoming and outgoing calls from both the IP and PSTN networks. IP-Tone features Voice Mail (unified or non-unified), Caller ID, Call Forwarding, 3 Way Conferencing, Follow Me, Assignment of Virtual/Real DID Numbers, as well as, Account & Mailbox Management via the Web. Dial-Gate with IP-Tone provides scalability and flexibility to configure the system to individual customer needs.
“An obstacle that many service providers of VoIP are striving to overcome, is attempting to reach customers (both residential and SMEs) sitting behind a firewall at the far end. Networks that contain firewalls result in inconsistent call completion. More often than not, resulting in call failure”, said Mohamed El Mohri, Dialexia CTO. “We have added a module to our SIP server, which has enabled our Dial-Gate software to manage the far end firewall, ensuring that dependable and secure VoIP services can be delivered to the end customer”.
“The launch of IP-Tone, along with our existing Dial-Gate software, allows us to offer a complete solution to ITSPs, VARs, and OEMs, enabling them to deliver a full range of services to meet the needs of the SOHO (small office/home office) and residential markets”, said Roger Hobeishe, Dialexia Director of Sales.
For further information about Dialexia’s line of Dial® products, please visit http://www.dialexia.com/index.jsp.
About Dialexia Communications Inc.,
Dialexia Communication’s is a true pioneer in the world of IP Telephony and Call Processing. With satisfied customers world wide and a rich history of technological innovation, Dialexia offers a full suite of integrated IP Telephony products and ITSP solutions. Dialexia’s mission is to provide innovative and advanced Voice and data over IP switches for Small carrier ITSP, phone booth retail outlets and for VoIP telephone systems for SMEs. All Dialexia products are designed to work with the industries leading gateways, facilitating installation and compatibility. Dialexia Communication’s, is headquartered in Montreal, Canada. For more information call 1-514-421-1151 or visit www.dialexia.com
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Dialexia Releases Dial-Gate with IP-Tone
December 13, 2004
Montreal, Canada
Dialexia Corporation, a leading provider of VoIP Traffic Management and Billing Solutions for the ITSP, announced today the release of their new software Dial-Gate with IP-Tone® services. Dial-Gate® is a Web-based SIP proxy and a centralized routing server for both the VoIP & PSTN networks. Among its many features are; A Pre/Post Paid Billing System, Real Time Monitoring, and CDR Management. With the added option of IP-Tone, Dial-Gate can handle incoming and outgoing calls from both the IP and PSTN networks. IP-Tone features Voice Mail (unified or non-unified), Caller ID, Call Forwarding, 3 Way Conferencing, Follow Me, Assignment of Virtual/Real DID Numbers, as well as, Account & Mailbox Management via the Web. Dial-Gate with IP-Tone provides scalability and flexibility to configure the system to individual customer needs.
“An obstacle that many service providers of VoIP are striving to overcome, is attempting to reach customers (both residential and SMEs) sitting behind a firewall at the far end. Networks that contain firewalls result in inconsistent call completion. More often than not, resulting in call failure”, said Mohamed El Mohri, Dialexia CTO. “We have added a module to our SIP server, which has enabled our Dial-Gate software to manage the far end firewall, ensuring that dependable and secure VoIP services can be delivered to the end customer”.
“The launch of IP-Tone, along with our existing Dial-Gate software, allows us to offer a complete solution to ITSPs, VARs, and OEMs, enabling them to deliver a full range of services to meet the needs of the SOHO (small office/home office) and residential markets”, said Roger Hobeishe, Dialexia Director of Sales.
For further information about Dialexia’s line of Dial® products, please visit http://www.dialexia.com/index.jsp.
About Dialexia Communications Inc.,
Dialexia Communication’s is a true pioneer in the world of IP Telephony and Call Processing. With satisfied customers world wide and a rich history of technological innovation, Dialexia offers a full suite of integrated IP Telephony products and ITSP solutions. Dialexia’s mission is to provide innovative and advanced Voice and data over IP switches for Small carrier ITSP, phone booth retail outlets and for VoIP telephone systems for SMEs. All Dialexia products are designed to work with the industries leading gateways, facilitating installation and compatibility. Dialexia Communication’s, is headquartered in Montreal, Canada. For more information call 1-514-421-1151 or visit www.dialexia.com