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Wed 01 of Aug, 2007 [09:27 UTC]

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  • 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
Server Stats
  • Execution time: 0.22s
  • Memory usage: 2.17MB
  • Database queries: 31
  • GZIP: Disabled
  • Server load: 2.32

SIPMerger Server - SIP Protocol Converter

SIPMerge™ Server

SIPMerge Server is a SIP protocol converter designed for bridging SIP networks and allowing full SIP header manipulation.

SIP, as an evolving protocol, often causes interoperability issues between different vendors. Using SIPMerge, these differences are overcome by providing a stateful proxy that remains in the call path ensuring compatibility between servers, gateways and IP-PBXs

SIPMerge Server is ideally suited for Microsoft Exchange 2007 Unified Messaging (UM) Server integrations. SIP traffic from the UM Server is sent over TCP/IP unlike the majority of IP-PBXs and VoIP gateways which use UDP.

Key features
• Transport Layer Conversion (TCP/UDP)
• Payload Header Manipulation
• Security
• Load-Balancing
• Fault Tolerance
• Windows based remote configuration, reporting and debug

Payload Header Manipulation
Extra fields can be added to the SIP header if necessary to comply with particular switches or end-points including Microsoft LCS/OCS and Exchange UM

Security
SIPMerge™ provides interoperability between different security and authentication regimes. Includes support for IPSEC

Load-Balancing
Various call distribution algorithms are supported including round-robin and performance based rules. Multiple network interface cards can be configured to support high volumes of SIP traffic and ease network congestion

http://www.postcti.com



Created by Gwen Nkambule, Last modification by Gwen Nkambule on Tue 01 of May, 2007 [15:39 UTC]

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