> I've been trying to find a solution that would allow our sip phones to
> communication with walkie talkies. Our setup is that we have sip phones
> setup in 2 locations, headquarters and dome. We can communication from
> headquarters and dome through sip phones, but within the dome we have
> technicians that use walkie talkies to communicate as they go about
> their work. Our hope is to allow 2 way communications from our sip
> phones at headquarters (or within the dome) with our technicians using
I've been trying to find a solution that would allow our sip phones to
communication with walkie talkies. Our setup is that we have sip phones
setup in 2 locations, headquarters and dome. We can communication from
headquarters and dome through sip phones, but within the dome we have
technicians that use walkie talkies to communicate as they go about
their work. Our hope is to allow 2 way communications from our sip
phones at headquarters (or within the dome) with our technicians using
their walkie talkies as they are working…
How can I find out One way latency from my PBX to my SIP Trunk Provider.
My SIP provider recommends a One way latency of 100ms for good Voice
quality. Ping request to their IP Address gives me a response in approx.
Will that be good enough for a SIP Trunk. Please help. We are trying to sign up with a new SIP Provider. Thanks,
the wav sound files that are created by using MixMonitor()-command are
not playable with Windows Media Player. I can play them with vlc-player and on my Fedora with Totem. This is one of the files : /var/ftp/104/2011-11-30_11:54:39_89000404.wav: RIFF (little-endian)
data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
What would be missing on the system (Centos 5.7) that makes wav-files
difficult for Windows Media Player ?
On one location, I've got from time to time (let say one a week) the
following issue :
the phone SoundPoint 650 works ok (can call or answer, display and sound
the sidecar looses its display : entries on sidecar's LCD screen are not
displayed anymore, or names are truncated, or BLF are not shown or updated. I only have one SPIP650 on this system so I can't compare with others. What could be the root cause of this ? Regards
I am using my first PAP2 device from linksys. Used many polycom phones...
I configured the PAP2 device with asterisk. I have the registration,
thought I was good to go.
Plugged in my Valcom 2924 public address analog connection, called the
and I got busy... very strange I thought. I then looked at the status page of the PAP2 and it says the following
Reg online and hook state OFF. How do I get the hookstate to be ON so that I can call into the device? I have power cycled it and it…
I know it's might not the right way to asking such stupid question. But I
want to take help from experts into VoIP fields so I have to decided to
post here. Please help me which will be the best VoIP conferencing phone which will
cover 10 Persians into conferencing with best audio support ?