I have a client using the Grandstream phones (not sure which model but it looks
fairly low-end) and they're lukewarm on them. The display doesn't tilt up for
easy viewing and the sound quality on the speaker phone leaves something to be
desired apparently.
But as basic, inexpensive, Asterisk
AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI
(single span). I'm sure I just have something goofed up in the
dialplans? I have a bunch of Polycom 331 IP phones connecting to the
server. I can dial the other extensions in the system fine and I can
dial long distance
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Wednesday, October 07, 2009 3:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Can dial long distance but not local?
AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI
(single span). I'm sure
] Can dial long distance but not local?
Ben Schorr wrote:
AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI
(single span). I'm sure I just have something goofed up in the
dialplans? I have a bunch of Polycom 331 IP phones connecting to the
server. I can dial the other
O.K., so AsteriskNow 1.4.26.2, FreePBX 2.6.0RC2.1 We have a Digium
TE205P connected to a single span if ISDN PRI. The Telco has assigned
us two local numbers to test incoming calls. I created an inbound route
for one of those DID's and assigned it to one of our extensions. Sounds
simple
-Original Message-
Too simple, apparently, when I dial the number the caller gets a
recording that it's a non-working number and this is what I see in
the
CLI:
Extension '8085255935' in context 'default' from '808xxx' does
not
exist. Rejecting call on channel 0/1, span 1
: Inbound call routing
On Fri, Oct 09, 2009 at 06:15:43AM -1000, Ben Schorr wrote:
-Original Message-
Too simple, apparently, when I dial the number the caller gets a
recording that it's a non-working number and this is what I see
in
the
CLI:
Extension '8085255935
Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones.
I've got G.729 loaded in the modules on the Asterisk server and on the
Polycom phones I've set G.729 to be the first preference of codec, but
still when I go SIP SHOW CHANNELS during active calls it still shows
(ULAW) (G.711) as the codec in
, December 15, 2009 8:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote:
Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones.
I've got G.729 loaded in the modules
only need a reboot for DAHDI changes (not always then...)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
Sent: Tuesday, December 15, 2009 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial
Of j...@jeff.net
Sent: Tuesday, December 15, 2009 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote:
Ahhh...yes, I think that may have been it. I moved G.729 to the top
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote:
O.K., interestingly enough when I call our extensions from my mobile
phone it still seems to be using ULAW, but when they dial out it
seems
] On Behalf Of Ben
Schorr
Sent: Tuesday, December 15, 2009 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
Sorry, I think I may have misspoke...
What I'm hoping for is that all of the connections between my phones
provider like (but not)
bandwidth.com? By eliminating the PRI element, you should completely
resolve the problem.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
Sent: Tuesday, December 15, 2009 2
-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Tuesday, December 15, 2009 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
- Ben
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
Sent: Tuesday, December 15, 2009 2:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
I thought I already did that - which is how they now get some
...@lists.digium.com] On Behalf Of Ben
Schorr
Sent: Tuesday, December 15, 2009 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
Sorry, I think I may have misspoke...
What I'm hoping for is that all
Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Tuesday, December 15, 2009 11:16 AM
To: Asterisk Users Mailing List - Non
all the
time.
I'm sure there is a way to do this through the freepbx gui, but like I
said, I
have no experience with freepbx.
-Dave
Ben Schorr wrote:
O.K., I think I'm catching on. I only have a single SIP.CONF file
that ALL of the extensions are using so I'm gathering that I need
: Re: [asterisk-users] Can't get G.729 to work...
Ben Schorr wrote:
O.K., I restored the Allow=ulaw in the sip_general_additional.conf
file, then I found the individual extension settings in the
sip_additional.conf file and I added
I would not go editing the individual files if you
I think Astlinux comes in under 100MB.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
Twitter: http://www.twitter.com/bschorr
Facebook: http://www.facebook.com/rolandschorr
-Original
Please see http://www.officeforlawyers.com/howask.htm
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com http://www.rolandschorr.com/
b...@rolandschorr.com mailto:b...@rolandschorr.com
From:
Is there some reason why I keep getting this same message from cool
dude over and over and over? And under different subject lines?
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com http://www.rolandschorr.com/
We deployed a single phone handset (Polycom 331) at a remote site. We
have a IPSEC VPN running between the firewall at the remote site and the
firewall at the site where our Asterisk/FreePBX box lives. We have used
a similar configuration for this site before and it worked fine.
We gave the
11:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX
On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr b...@rolandschorr.com wrote:
We gave the phone a static IP address and pointed it to the
configuration
We're using FreePBX 2.8 and there is a Reports tab but it doesn't seem
to actually do anything. Is there some secret/trick to getting a report
out of it that will tell us which extensions are placing calls? I've
tried every query on the form that I can think of. Is the reporting
disabled by
On 30/12/2010, at 8:36 PM, Ben Schorr wrote:
We're using FreePBX 2.8 and there is a Reports tab but it doesn't seem
to actually do anything. Is there some secret/trick to getting a report
out of it that will tell us which extensions are placing calls? I've
tried every query on the form
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