An FXO port needs to be connected to dial tone or your PSTN line. And
an FXS port needs to be connected to the station equipment(ie. a
physical phone).
The TDM410 is basically a channel bank to Asterisk, so the channel type
inside Asterisk is FXO to talk to the physical FXS card and FXS to ta
> >> You have hardware echo canceling *outside* of your T1 card?
> >
> > No, on the card.
>
> Then you definitely don't want 'echocancel=no' set, or you'll disable it.
When I thought that it was echo cancellers fighting each other, that's
exactly what I wanted to do.
--
_
This is a Polycom question, not an Asterisk question.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Tuesday, June 19, 2012 1:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
I'm trying to figure out how to change the redial, thus far if I hit redial
it will redial the last called I made that was answered, not the last call I
made that was not answer.
I'm using Asterisk 1.8
Thanks,
Motty
--
_
Am 19.06.2012 11:53, schrieb [Digital^Dude] ®:
> Machine specs: CentOS release 5.5 (Final)
> RAM: 4 GB
> CPU: Dual Xeon 2.66 GHz
> Asterisk 1.8.7.1 built by root on a x86_64 running Linux.
>
> *CLI> ulimit core
> Core file size (core) is effectively unlimited.
> *CLI> ulimit data
> Program data
Hello.
First sorry for my English.
I`m from Moscow, Russia.
I have a problem with getting caller id.
Dist: Elastix version 2.0.0.-36, Asterisk version 1.6.2.10.
There is a digium card TDM2400P in the server. The analog phone line
directly connected in this card. There is activated service "Euro Ca
It seems now that 1.8.13 now supports IMAP voicemail.
I just installed a fresh copy of PBX In a Flash and opted to install Asterisk
1.8.13. It seems now that when built with 'make menuselect' (after downloading
and making the IMAP client from UW and running './configure
--with-imap=/usr/src/
>> Interestingly, that isn't completely true. If it goes out a SIP trunk
>> to PSTN, it works fine, but when it goes out a SIP trunk to the SV8300
>> (where the T1 goes), it has the same problem. This was leading me to
>> believe that the problem was on the 8300.
>Well, that doesn't disprove m
On 06/19/2012 04:23 AM, Richard Kenner wrote:
You have hardware echo canceling *outside* of your T1 card?
No, on the card.
Then you definitely don't want 'echocancel=no' set, or you'll disable it.
The DAHDI layer has some buffering that can help with jitter, but the
default buffers can on
Is there a way to have ALSA accept more than one
incoming call?
I have asterisk running a box with an audio source input.
So the incoming call just connects the audio feed.
Issue is I want "at times" to source that feed to more than
one call.
Can I do that? How is it accomplished?
Thanks,
Jer
Machine specs: CentOS release 5.5 (Final)
RAM: 4 GB
CPU: Dual Xeon 2.66 GHz
Asterisk 1.8.7.1 built by root on a x86_64 running Linux.
*CLI> ulimit core
Core file size (core) is effectively unlimited.
*CLI> ulimit data
Program data segment (data) is effectively unlimited.
*CLI> ulimit descriptors
*
Hello,
I short question:
I want to connect Asterisk to OpenBSC with mISDN, mISDNuser and LCR.
Do I need chan_lcr?
I have:
Asterisk 1.8
mISDN .v2 integrated in Kernel 3.0.22
mISDNuser
lcr 1.7
HFC-E1 Evaluation board from cologne chip
I tried to configure Asterisk with <./configure --prefix=/us
> You have hardware echo canceling *outside* of your T1 card?
No, on the card.
> The DAHDI layer has some buffering that can help with jitter, but the
> default buffers can only handle 80ms of jitter. You can increase this by
> setting the 'buffers' option in chan_dahdi.conf; each buffer is 20
Am 18.06.2012 21:49, schrieb James Sharp:
On 6/18/2012 11:52 AM, Thorsten Göllner wrote:
Hi,
I am trying now for over 4 hours setting up cdr-logging via odbc into a
mysql database. But with no success. Do you have any hint for me?
*SNIP*
But after a call hangup I get the following error:
c
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