Zaptel seems to be running.
Channel status:
Channel: 4
File Descriptor: 13
Span: 1
Extension:
Dialing: no
Context: from-pstn
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner: Zap/4-1
Real: Zap/4-1
Callwait:
Threeway:
Confno: -1
Propaga
Hello,
Please Confirm if the dahdi/Zaptel service is running .
check your channels status.
On Wed, Mar 24, 2010 at 9:29 AM, Balu Raman wrote:
> I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls
> are not being picked up. I don't find anything unusual in asterisk
> log. I
I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls
are not being picked up. I don't find anything unusual in asterisk
log. I am clueless where I should look. I also find
zapata-additional.conf empty. The trouble started when the system was
accidentally shut down and rebooted.
A
At 07:52 AM 2/26/2010, you wrote:
>Outgoing minutes (depending where you are in the world) should cost less
>than US$0.02 per minute with no monthly standing charge.
I'd not been paying attention and was recently surprised to find that
my rates have dipped under .01/minute for prepay at $20 every
SCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
From: Aditya Kumar
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Fri, February 26, 2010 11:08:52 AM
Subject: Re: [asterisk-users] : PSTN calls
TH
On Fri, 26 Feb 2010, Aditya Kumar wrote:
> I understand about FXO, FXS.
>
> I want to have a connection from my Asterisk box to the External ptstn
> world via My home phone line. So, if I use USBfxo, Can I connect it
> directly to my wall socket?
>
> so set up will be : Asterix linux Box--usb ha
.@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
From: Aditya Kumar
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Fri, February 26, 2010 10:05:26 AM
Subject: Re: [asterisk-users] : PSTN calls
On Fri, 26 Feb 2010, Aditya Kumar wrote:
> sip-xlite talking to Asterisk Asterisk Box is connected to USBfxo
> (example). USBfxo connects to my Phone line.(which is fxo)
>
> so with translations defined I can call, from Analong phone to sip lite
> (internally)
>
> Now if I want to make calls to
From: Steve Edwards
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Fri, February 26, 2010 9:15:19 AM
Subject: Re: [asterisk-users] : PSTN calls
Un-top-posting...
> On Fri, 26 Feb 2010, Aditya Kumar wrote:
>
>> Now I want to make call from sipx-lite to PSTN using
Un-top-posting...
> On Fri, 26 Feb 2010, Aditya Kumar wrote:
>
>> Now I want to make call from sipx-lite to PSTN using asterisk. can any
>> please suggest me which Hardware card that I can buy?
> From: Steve Edwards
>
> As an alternative, you can get an account with a SIP or IAX termination
>
: Fri, February 26, 2010 7:52:04 AM
Subject: Re: [asterisk-users] : PSTN calls
On Fri, 26 Feb 2010, Aditya Kumar wrote:
> Now I want to make call from sipx-lite to PSTN using asterisk. can any
> please suggest me which Hardware card that I can buy?
As an alternative, you can get an account
On Fri, 26 Feb 2010, Aditya Kumar wrote:
> Now I want to make call from sipx-lite to PSTN using asterisk. can any
> please suggest me which Hardware card that I can buy?
As an alternative, you can get an account with a SIP or IAX termination
provider.
Outgoing minutes (depending where you are
On Fri, Feb 26, 2010 at 2:35 PM, Aditya Kumar wrote:
> can any please suggest me which Hardware card that I can buy? and use ( pl
> give me all ths list of cards which are good.).
Here is a starting list of Asterisk hardware
http://bit.ly/a6yX6h
--
_
Hi All,
I have installed astriesk 6 and am able to make calls using sip x-lite.
Its working as I expected.
Now I want to make call from sipx-lite to PSTN using asterisk.
can any please suggest me which Hardware card that I can buy? and use ( pl give
me all ths list of cards which are good.
On Fri, Feb 16, 2007 at 12:39:54AM -0500, Allen Casteran wrote:
> We have SIP phones connecting to *, and our PSTN lines connecting
> through an Astribank FXO.
>
> Internal Sip<->SIP calls are clear.
> External calls through the Astribank get occassional low level buzzing
> for about 1/2 - 1 sec
We have SIP phones connecting to *, and our PSTN lines connecting
through an Astribank FXO.
Internal Sip<->SIP calls are clear.
External calls through the Astribank get occassional low level buzzing
for about 1/2 - 1 sec at a time.
Those are usually accompanied by some light pops or static
N
I just did it by ear. Got it right in less than 5 minutes.
--- Paul Goodyear <[EMAIL PROTECTED]> wrote:
> I thought the txgain, rxgain was purely for echo settings.
>
> Is there a rough guide to this process, or is it a simple case of
> changing values and testing them?
>
> Thanks.
>
> On 9
I thought the txgain, rxgain was purely for echo settings.
Is there a rough guide to this process, or is it a simple case of
changing values and testing them?
Thanks.
On 9/15/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hi Paul
>
> There are two settings in zapata.conf called txgain and r
Hi Paul
There are two settings in zapata.conf called txgain 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
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 us,
Hi,
Asterisk either need to know when the remote caller ends his call,
or it must detect the silence.
Simplest solution is to activate silence detection, see
voicemail.conf.
You may need to do some testing to get the proper
"silencethreshold" setting.
Also search the archive, this is a often
Hi all,
In my Asterisk setup, incoming calls through
Cisco PSTN gateway to Asterisk extensions sounds work fine. All calls can be
terminated properly after hangup. However, when calls were forwarded to
voicemail, after recording & hangup the PSTN calls and cisco FXO
port remained connected
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