I am trying to set up a Grandstream GXV-3000 Video
phone to Asterisk ver 1.2.21.1. The problem I'm
having is that it can call other SIP phones, but not
vice versa. Can someone tell me where is the problem?
TIA!
Here's part of my configurations:
--
sip.conf
--
; 113 is the
'disallow=all'
to [113]'s
definition, just before the 'allow=h263' and the
'allow=ulaw' i'm
suggesting. If, theoretically, friend 113 will ONLY
use h263, does
X-Lite support this codec?
Have you tried kicking the verbose level at the
console up a little bit?
Moj
hin lee wrote:
I am
I've installed AsteriskNow version 1.0.2 on a test machine and is trying to
configure my Auto-Provision over http. I spent hours trying to figure out why
it wasn't working and finally realize that my files are not displaying under
the phoneprov directory. I tested by putting a test html file
Ed,
Sounds like the digitmap on the Polycom phones is the issue. You can read more
about the digitmap from:
http://www.voip-info.org/wiki/view/Polycom+Phones
Digitmap reference
Example: [2-9]11|0T|011xxx.T|91[2-9]x|[1-8]xx
It means the following:
* [2-9]11: 911 rule: x11 are
Hi,
A newbie here trying to learn Asterisk. I've installed PiAF v.1.3(PBX in A
Flash) and trying to set up the TDM808E card as a test. For now I only have
one analog line. I went into the FreePBX interface and created a ZAP trunk
with 1 as the Zap Identifier.
When I try to call out, I
/interfacing-to-a-pstn
Hope this will help the next person who may encounter this issue.
--- On Tue, 10/28/08, hin lee [EMAIL PROTECTED] wrote:
From: hin lee [EMAIL PROTECTED]
Subject: [asterisk-users] CHANUNAVAIL with a TDM800 card
To: Asterisk Users Mailing List - Non-Commercial Discussion
: Re: [asterisk-users] CHANUNAVAIL with a TDM800 card
To: asterisk-users@lists.digium.com
Date: Thursday, October 30, 2008, 11:20 AM
On Thu, Oct 30, 2008 at 11:03:03AM -0700, hin lee wrote:
I got this working. For what it's worth,
here's what the issue.
The channel wasn't getting created
: Re: [asterisk-users] CHANUNAVAIL with a TDM800 card
To: asterisk-users@lists.digium.com
Date: Thursday, October 30, 2008, 1:06 PM
On Thu, Oct 30, 2008 at 12:56:06PM -0700, hin lee wrote:
Tzafrir,
You are correct! I didn't have to commented out
the unused FXO ports. So to revise my earlier
Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't
figure out why the volume is so low. How can I adjust the volume control on
Asterisk? It's at max on the handset phones.
Thanks!
Hin
___
-- Bandwidth and
firmware is used on the
phones? Are
you calling from one phone to the other?
Darrick
Michael Graves wrote:
Probably has nothing to do with Asterisk. You can set
the volume and
persistence in the phones config files.
Michael
On Fri, 14 Nov 2008 22:43:45 -0800 (PST), hin lee
=dggrkn86_5ss94tkf6hl=en
Phone cfg file
http://docs.google.com/Doc?id=dggrkn86_4csdzthf9hl=en
Server cfg file
http://docs.google.com/Doc?id=dggrkn86_6gp7hr9fghl=en
SIP cfg file
http://docs.google.com/Doc?id=dggrkn86_3djzb86d7hl=en
--- On Sat, 11/15/08, hin lee [EMAIL PROTECTED] wrote:
From
Anybody know why the volume on calls are so low? How can I increase the volume?
--- On Sat, 11/15/08, hin lee [EMAIL PROTECTED] wrote:
From: hin lee [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Polycom low volume
To: Asterisk Users asterisk-users@lists.digium.com
Date: Saturday
Is there a way to have Asterisk talk to a Avaya IP
Office phone system? If it's possible, where can I
find the instructions?
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
Whenever I try to drag calls to the Parking Lot or On Hold, FOP would
drop my calls. I have searched online and have found
similar problem, such as the link below. I have tried their solution
but still the FOP is not working correctly. I even installed the
HUDLite server and is getting the same
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Mon, October 5, 2009 2:40:50 AM
Subject: Re: [asterisk-users] Drop calls when using Flash Operator Panel
hin lee wrote:
Whenever I try to drag calls to the Parking Lot or On Hold, FOP would
drop my
I got a few newbie questions.
If I get an echo cancellation module for my Digium TE121 card, will I need to
do any adjustments/configuration in Asterisk? Is the hardware better than the
software version?
TIA!
___
-- Bandwidth and Colocation
I am having echo issues on our Asterisk box using a PRI circuit. I was using
the software echo cancellation and that helped a bit but didn't solve it
completely. So I went and bought a Digium echo cancellation module for the
TE121 card. That made it even worst, getting more echo on external
The echo between our extensions (using Polycom 550 handsets) disappears once I
removed the Digium echo module. We are still experiencing some echo on land
line calls, using dahdi to connect to our PRI circuit.
What kind of settings do you recommend for the txgain and rxgain? Do I make
the
I had once considered Switchvox b/c of the user interface simplicity. However
with the lack support of H323, I had to look into a different solution. I am
currently on Elastix, which is a great app! Everything is done using the web
interface, eg, network configuration, hardware, phone
Processor, 2 GB DDR
Memory, 160 GB 7200 rpm Hard Drive)
From: Mike mikef1...@gmail.com
To: hin lee hi...@yahoo.com
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, December 10, 2009 9:43:05 AM
Subject: Re
The echo between our extensions (using Polycom 550 handsets) disappears
once I removed the Digium echo module.
Are you routing internal calls from SIP - DAHDI - SIP? The digium
echo module will not have any effect on pure SIP - SIP calls. Do
you have acoustic echo cancellation active on
If I installed a Digium echo cancellation module on my TE121 card, do I need to
remove the echocanceller line under the system.conf? How should I have it?
This is my system.conf:
bchan=1-23
dchan=24
echocanceller=mg2,1-23
Thank you!
Hin
From: hin lee hi
I am consider replacing my TDM card for a FXS gateway. Anyone has any issues
with the Grandstream GXW-4004 on Asterisk? I would like some feedback before I
spend the $$ this device.
http://www.voip-info.org/wiki/view/Grandstream+GXW-4004
Thanks!
yes, fxs for my fax machines.
From: Lyle Giese l...@lcrcomputer.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sat, January 2, 2010 3:29:13 PM
Subject: Re: [asterisk-users] Grandstream GXW-4004
hin lee
I have posted my problem on the link below, but didn't get any answer. I am
hoping someone here can help me with this issue. Here's my problem:
I am using H323 to talk between Asterisk and Avaya IP Office 500. For
some strange reason, when we are talking on a VoIP call, we get
disconnected
You can practice Asterisk using free SIP phones. This way you can call from
extension to extension.
SJ Phone
http://www.sjlabs.com/sjp.html
X Lite
http://www.counterpath.com/x-lite.html
From: UIT DEVELOPMENT uit...@gmail.com
To: Asterisk Users Mailing List
We are using Polycom 550 and 650 phones and OSLEC echo cancellation software
with Asterisk. Occasionally, we get echo on our PRI phone calls. The echo is
always from our voice echoing back to us. How can I fix this echo? I have
tried installing the VPMADT032 module on our TE121 card, but
You can also go with external FXO gateways, e.g. as AudioCodes Mediatrix, etc.
This way you can avoid IRQ issue with standard cards.
From: David Backeberg dbackeb...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
I am using H.323 to create a trunk between Asterisk and Avaya IP Office system.
Everything is working correctly, Asterisk
can call Avaya and vise versa. Now I create a conference room with a
user pin in Asterisk. Avaya can call into the conference room, but can
enter the pin number. The error
Beside the port number and the alaw, the only difference is the dtmf. I added
this into my ooh323.conf and it still didn't work.
dtmfcodec=127
dtmfmode=rfc2833
I also tried: dtmfmode=h245signal
This is to an Avaya IP Office 500.
--
Usually I see /DAHDI/*channel #*, but today I see this
AsyncGoto/DAHDI/*channel#* on one of my call. What does this mean?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
I have quite a number of users complaining that when they are using handsfree
to talk over a landline, the other end can't hear them. It's like the person
is speaking 5 feet away and can barely hear their voice. However between
internal SIP calls, it's fine.
What could be the problem?
I am also having this issue with the MOH. Would be nice to find a solution!
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Fri, January 29, 2010 3:43:12 PM
this, but the
assumption would be that your external calls are DAHDI based, so you might need
to tweak txgain in dahdi.conf.
From:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee
Sent: Friday, January 29, 2010
Using FreePBX, is there a way to play a beep sound when you are connected to a
parked
call? Right now, it's dead silence and we can't tell if the call has
been connected.
--
_
-- Bandwidth and Colocation Provided by
] parked calls
hin lee wrote:
Using FreePBX, is there a way to play a beep sound when you are
connected to a parked call? Right now, it's dead silence and we can't
tell if the call has been connected.
I don't know about FreePBX, but under the features.conf, there is:
courtesytone = local
Hi,
I have a strange problem with all of our Polycom 550 650 phones. I am
running a TFTP server on my Asterisk server and option 66 Boot Host pointing to
Asterisk on my DHCP server. The auto-provisioning is working because the
phones are registering correctly with their extension. If I
anyone?
From: hin lee hi...@yahoo.com
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Fri, March 12, 2010 10:08:53 AM
Subject: Polycom not updating the directory list
Hi,
I have a strange problem with all of our Polycom 550 650 phones. I am
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee
Sent: Thursday, 18 March 2010 4:56 PM
To: Asterisk Users
Subject: Re: [asterisk-users] Polycom not updating the directory list
anyone?
From: hin lee hi...@yahoo.com
To: Asterisk Users asterisk-users
With the price of FXS gateway, why not just get SIP phones? Polycom 330 is
around $60-$110 a piece.
From: mir shahnawaz shahnawaz...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tue, March 30,
Figured out my issue. My contacts are in mac-addr-directory.cfg when it
should be in mac-addr-directory.xml.
When did Polycom switched from CFG to XML?
From: hin lee hi...@yahoo.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
check the IRQ and make sure the TDM410P has it owns IRQ.
From: Danny Dias ing.diasda...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Fri, April 9, 2010 4:52:05 PM
Subject: [asterisk-users] Problems with Fax over TDM410P
Hello my friends...
We are
Can't upgrade the version. how about buying a FXS gateway and be done with
the issue. Go to ebay and search for AudioCodes. You can get 1 FXS port
gateway for around $30 to 2 FXS at $85. Probably the best bet is to convince
the customer to upgrade Asterisk.
I have a question about the blind transfer
using ##. This works great on our cordless phone, but there have been
occasions that we can't transfer using ##. I was able to reproduce the
issue by doing the following:
1) Call in from the outside line,
2) Ask the operator to transfer me to an
Thanks for replying Noah. I'm using FreePBX web interface and have a ring
group that rings 4 phones as the operator. I do know that the context type is
from-internal but when it rings as below, the context type becomes
from-pstn. Can you tell me where exactly to go and change in the FreePBX
I too, had really bad echoes on a TE121 w/ echo module. When I removed the
module, I haven't had as much echoes as before.
From: Sascha Ferley sascha.fer...@infineon.net
To: asterisk-users@lists.digium.com
Sent: Sun, May 16, 2010 12:00:31 PM
Subject:
I have discussed QoS with our ISP and in order to implement this, I need to
make
sure all VoIP packets are marked in the IP packet header (IPP bits?). Does
Asterisk automatically marks the VoIP packets or do I need to do something in
Asterisk? I need to do this for SIP and H323 protocols.
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