search for: mindspeed

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2004 Sep 06
3
SIP rtp port forcing
When a SIP call starts (INVITE / 200 OK), asterisk seems to create a random port number for voice (rtp) packets. Is it possible to force this port value (without using reinvite since i am trying to use SIP against something else than sip) thanks a lot in advance -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Apr 18
2
[Bridge] I want to RTFM -- so where's the FM?
...the managed switch. I'm hoping that it's as simple as writing a driver for the managed switch and registering it with the bridging code. Is there any documentation on the internals of the bridging software or do I have to paw through the source? Thanks! -- Wade Guthrie wade.guthrie@mindspeed.com
2006 May 10
1
mg3000-r fxo gateway provides more feature to work with asterisk
...MG3000-R could detect the voip quanlity, when voip line failed, it change to pstn line automaticly. 3. pstn caller number transfer. When pstn call in, the mg3000-R start voip call to the asterisk using pstn caller number instead of gateway number. 4. multy region pstn singal support. By using MindSpeed technology, it integrated many region's pstn singal. Why we choose a voip fxo gateway while not a asterisk card. 1. voice process is a realtime task. PC operate system is not a realtime one. Voip gateway use its own dsp to do voice process while asterisk card use PC CPU to do t...