Thanks, Steven.
Do both parties hear the crackling, etc?
No, just the users in the Office, and its like a background crackle that
happens with voice and silence.
This happens all the time, every call.
Called the Office number from VoIP phone to VoIP phone and the quality is
terrible.
Don't
Hi all,
I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and
1XFS modules.
The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM.
Sangoma A200 has 3 analogue PSTN lines connected.
This server is based in Office 1, with 5 users all with a Linksys SPA942
VoIP Handset.
There is
Update,
Still not sorted, I have checked some tools on the TrixBox and using
the wanrouter I was able to check the voltage on lines.
The three result are when there is no call active
[EMAIL PROTECTED] ~]# wanpipemon -i w1g1 -c astats -m 1
--- Voltage Status (FXO,port 0) ---
Voice work flawlessly
PaulG.
On Wed, Mar 19, 2008 at 8:17 AM, David Quinton [EMAIL PROTECTED] wrote:
On Tue, 18 Mar 2008 14:06:44 +, Paul Goodyear [EMAIL PROTECTED]
wrote:
Hi,
I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there
are 3 BT lines connected directly
Hi all,
I bought a Sangoma A200 card from an online supplier and explained
exactly what I wanted,
3 incoming phone lines to PBX and a life line (some where to connect a
standard BT phone to the PBX incase the power goes, making the BT
phone ring).
I was told to order
1 x FXS module (2 FXS
Excellent, thanks for that Tzafrir.
PaulG.
On Tue, Mar 18, 2008 at 1:18 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Mar 18, 2008 at 12:02:26PM +, Paul Goodyear wrote:
Hi all,
I bought a Sangoma A200 card from an online supplier and explained
exactly what I wanted
Hi,
I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there
are 3 BT lines connected directly to these ports.
One of the lines has BT FeatureLine Compact and this is the line I am
having problems with, the other 2 lines are working perfectly,
detecting CID, answering incoming calls
] wrote:
On Tue, 18 Mar 2008, Paul Goodyear wrote:
Hi,
I have a TrixBox install with a Sangoma A200 and 4 FXO ports, there
are 3 BT lines connected directly to these ports.
One of the lines has BT FeatureLine Compact and this is the line I am
having problems with, the other 2
PM, Ade Vickers
[EMAIL PROTECTED] wrote:
Paul Goodyear wrote:
I have had a BT phone plugged into these lines for about 3 week
prior to testing on asterisk, and all the lines are fine. Even
the first line, it rings and answers ok.
Apologies if this seems dumb, but have you done the swap
FIXED,
FYI:
Turning the secret OFF in the incoming context fixed the problem.
On 10/10/05, Paul Goodyear [EMAIL PROTECTED] wrote:
Hi all.
Just as a quote note, can I thank everyone on this list. I find my
self finding pretty much every answer I am looking for on here. And a
big thanks
Hi all.
Just as a quote note, can I thank everyone on this list. I find my
self finding pretty much every answer I am looking for on here. And a
big thanks to all thoughs helping us out. Mass Respect :)
Ok, I'm using a SIP provider (SipGate UK) to do my international
dialing etc, working great
I dont want to start a RTFM thread, but can someone jsut clear this up for me.
In zapata.conf I have
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=incoming
signaling=v23
rxwink=300 ; Atlas seems to use long (250ms) winks
Sip to sip calls are fine, both local on Asterisk and over a SIP
gateway, however some people who call on the PSTN line say we are very
queit and vice versa, can the volume be turned up on the PSTN line?
The volume buttons on the VoIP phones only turns up the others voice,
so this is a fix for
and rxgain. You can set
these to adjust the volume on your PSTN lines. They can be set in db or as a
percentage.
Garth
--- Paul Goodyear [EMAIL PROTECTED] wrote:
Sip to sip calls are fine, both local on Asterisk and over a SIP
gateway, however some people who call on the PSTN line say we
I am on UK Cable (Telewest) will this callerid patch work with the
cable caller id?
Thanks.
Paul.
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Sorry about my previous post, outbound routing this was clearly
available in AMP.
However,
How would I go about making all calls from PSTN line 1 (X100P #1) ring
call group #1, PSTN line 2 (X100P #2) call group #2, and the SIP
incoming calls route to group #1 also. Is this done via dial plans?
Sorry about my previous post, outbound routing this was clearly
available in AMP.
However,
How would I go about making all calls from PSTN line 1 (X100P #1) ring
call group #1, PSTN line 2 (X100P #2) call group #2, and the SIP
incoming calls route to group #1 also. Is this done via dial plans?
Current setup
2 x X100P cards connected to 2 analogue lines
Using prefix 7 and 8 before number
SIP gateway to SipGate to make VoIP calls
Using prefix 9 before number.
Is it possible so that if I dial a number:
0800 8000 8000
that it will try to route the call over the first analogue line, if
Is the X100P FXO PCI Card capable of detecting a fax, answering the
call, and then emailing the fax content to an email address?
Thanks.
Paul.
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Asterisk-Users@lists.digium.com
On 7/4/05, Rich Adamson [EMAIL PROTECTED] wrote:
Is the X100P FXO PCI Card capable of detecting a fax, answering the
call, and then emailing the fax content to an email address?
I haven't tested the digium x100p for several months, but I believe
it has the same issue as the TDM card
You should set Send DTMF to via RTP (RFC2833).
It works for me. I'm running firmware 1.0.6.3
Have you set Send DTMF to via SIP INFO
Thanks, set to RFC2833 and works a treat, thanks.
Paul.
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Hi,
Just playing with a couple of Budgetone 102 phones and they are pretty
good for the price.
The only problem i'm having at the moment is when I get a voicemail on
the Asterisk box the LCD flashes.
Dialing
*98 goes to the VoiceMail Manager, and asks for mailbox, I enter 201,
then asks for
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
8|. to place all
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
8|. to place all
Hello, I've been looking at the DialPlans by some poeple using
Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to
create Outbound routes etc, does using the web admin give the same
effects?
When I add a SIP Trunk with my Sipgate settings and use a pattern of
8|. to place all
that Transfer option in the Dialplan
i.e.
exten = 301,1,Macro(stdexten,301,${NATE})|Ttr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul Goodyear
Sent: Wednesday, April 20, 2005 9:26 AM
To: Asterisk-Users@lists.digium.com
When I place a call on my softphone to a external number the call is
placed, when I click transfer, dial internal extrention (e.g. 202)
then hit transfer again, the call is transfered to the 202 extention
fine.
However, when the other way Internal call comes in, extension 201
answers, and
When I place a call on my softphone to a external number the call is
placed, when I click transfer, dial internal extrention (e.g. 202)
then hit transfer again, the call is transfered to the 202 extention
fine.
However, when the other way Internal call comes in, extension 201
answers, and
Yes all ports have been forwarded on the iptables section at top
UDP/5060, UDP/4569, UDP/5036, UDP/1:2, UDP/2727
Doing a simple telnet to these ports non of them are open, even from
inside the LAN, so the issue is on the asterisk box rather than the
forwarding I think.
On Wed, 23 Mar
ah, you learn something every day :)
I will have a look at the asterisk conf files next week, thank for the info.
Paul.
On Thu, 24 Mar 2005 11:13:57 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] wrote:
On Thu, 24 Mar 2005, Paul Goodyear wrote:
Yes all ports have been forwarded
Asterisk only uses UDP, and AFAIK, you also need UDP ports 1 to
xx see /etc/asterisk/sip.conf for details.
You will also need to set the various NAT related config options in the
sip.cfg file.
As far as getting it to work on your LAN, well, I though you said you
had X-Lite working for
I'm all up for reading and looking round for people in the same boat
to try and solve the issue together, but there appears to not be large
community yet, just the asterisk mail lists.
I got Asterisk working with X-Lite great now for internal calls and
also calling land line numbers etc. The two
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