Displaying 6 results from an estimated 6 matches for "oneringnetworks".
2006 Feb 05
11
TE411P Really Bad Echo
...ect=no
busycount=7
channel=>98
;;;;;;[111]
signalling=fxo_ks
record_out=Adhoc
record_in=Adhoc
mailbox=111@device
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callerid=device <111>
busydetect=no
busycount=7
channel=>97
Thanks
Stagg
www.oneringnetworks.com
2005 Jun 19
4
Polycom 500 Sound Problem
Hi all,
I've been messing around with the g729 codec in some phones I use and had made all phones use the codec for all calls for testing purposes. The problem is when I attempt to dial out on my Polycom IP 500 (test happens to be calling my cell phone) I can only hear sound coming one way, I recieve nothing from one user, just silence, yet I can talk one way perfectly. Now I tried the same
2005 Jun 17
5
Presence and IM?
We have been running Asterisk for about a month now and one of the things I
miss the most is the ability to se who's online and available and who's not.
Surely, there's the manager interface, but unless you'd want to install
extra software on each client computer, this is not a good option.
Then there's the presence / IM function in SIP. Since we're only using SIP
2006 Feb 21
3
Recommended rack-mountable server anyone?
Hey everyone,
I've been doing a lot of research into a decent server for Asterisk
but I seem to be running and circles and now I am turning to you. The
issue I have is it needs to be rack mountable (so a Dell SC430 isn't
going to work) and preferably have 3 pci ports. The problem that I
seem to be running into is that when I look at servers from Dell or
IBM or the like they only seem to
2005 Jun 16
1
Routing SIP to Cisco routers running IOS 12.3+
I've experiencing some difficulty passing inbound calls from the PSTN,
through a large Asterisk switch and down our network to a Cisco 1751
router. This router has 4 FXS ports and is running IOS 12.3.
Outbound dialing from phones on the FXS ports of the router works
flawlessly, but inbound calls fail as though the Asterisk server does
not see the extensions representing the FXS ports as
2006 Jan 25
0
Echo while using Headset with Polycom IP 501 / 601 Asterisk 1.2.1
I'm hearing an echo when using a headset with my IP 501 / 601. The
phones are using BR 3.1.2 and SIP 1.6.3. I use tftp to configure the
phones. The sip.cfg is the default from polycom except for the
parameters required to connect the phones to asterisk.
I have absolutly no echo with the handset, but do have a slight echo on
the speaker phone. I haven't ruled out room acoustics as the