It sounds to me like you should first discuss it with adtran. The standard echo
cancellation for Asterisk have a hard time cancellingecho generated at the far
end, especially if the echo tail/delay is notminimal.
If adtran can not solve the problem at their end, you can use a server-side
echo
I have problem with offer SDP that firefox nightly generates. It writes out the
following error on asterisk:
WARNING[25424][C-0004]: chan_sip.c:10936 process_sdp_a_dtls: Unsupported
fingerprint hash type 'sha-2' received on dialog '2457893540'
SDP:
v=0
o=Mozilla-SIPUA 14911 0 IN IP4 xxx
s=SI
According to this:
https://wiki.asterisk.org/wiki/display/AST/Video+Telephony
yes.
I have a local server with two video phones - running SIP to each phone.
Works.
Then I have an IAX2 connection from that local machine to another machine.
then a SIP connection from that machine to another mac
On Mon, 7 Jan 2013, bilal ghayyad wrote:
Thanks for the help and it seems I deleted some of my emails by mistake!
I am sorry if I repeated my question.
The lists are archived at:
http://lists.digium.com/pipermail/asterisk-users/
On Wed, 2 Jan 2013, bilal ghayyad wrote:
As I see that
Hi,
2013/1/7 Doug Lytle :
> I'm looking for suggestions on a IP based amp or similar that could drive
> the current speakers? I was envisioning a unit that would register as a SIP
> extension then would handle auto-answer that I could send a sound file to.
I suggest you to take a look on snom P
Whether you use .call file or AMI, you should still do the call/page using a
context and that context run the PHP script.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Monday, January 07, 2
Thanks for the help and it seems I deleted some of my emails by mistake ! I am
sorry if I repeated my question.
As I see that the call file is used to generate calls, can I use this technique
to page the Phones?
It is one wave file only that need to be Paged for all the Phones connected on
the
I have several adtran 624 with 24 FXS ports hooked up to analog phones. The
adtran is connected to asterisk via a channelized T1 into a digium TE820. I
have hardware echo canceling enabled on all channels/spans, but there is still
echo on the lines for both calls out of the trunk, as well as
- Original Message -
> From: "Logan Bibby"
> Does anyone have a good contact for their sales? I've attempted
> calling their Enterprise sales a few times and was just spun around
> in circles. Having a sales rep I can just call would be awesome.
Logan,
We have an account manager that
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, January 07, 2013 3:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 support of video
Does
Does IAX2 support a video call ?
Jerry
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
On 01/07/2013 03:41 PM, Doug Lytle wrote:
The blowing fuses could be related to spikes etc., from a poor
connection to the source, or a problem with the source hardware.
>> If the amps are good, you could just drive them from a cheap phone
with a regular headset jack
They aren't, seem to b
>> If the amps are good, you could just drive them from a cheap phone with a
>> regular headset jack
They aren't, seem to be blowing fuses more often.
Thanks!
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
>> you might take a look at Valcom's products. http://www.valcom.com
I've bookmarked the page,
Thanks!
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
--
>> the speakers could probably be adapted to work off of any SIP phone
>> headset/handset
Interesting thought, but the amp needs to be replaced as well. The 10 year
Asterisk system will most likely be replaced with a Dell 1U and a Dual-Port
PRI.
Thanks for the suggestion!
Doug
--
Ben
On Mon, Jan 7, 2013 at 1:10 PM, Doug Lytle wrote:
> We currently have an Asterisk system that is hooked up to our old paging
> speakers via sound card, plugged into two amps.
>
> Each amp drives up to 8 analog speakers in each warehouse (we have 2).
> Both warehouses are around 30k square feet.
If you want something a little more enterprise ready and tested than a
RaspberryPi, you might take a look at Valcom's products.
http://www.valcom.com
We use them for our paging and have been fairly happy with them. Only had
one small issue that a firmware upgrade took care of.
Kevin Larsen - S
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 07, 2013 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Paging unit suggestions
We currently have an Aster
On Mon, Jan 7, 2013 at 2:10 PM, Doug Lytle wrote:
> I'm looking for suggestions on a IP based amp or similar that could drive
> the current speakers? I was envisioning a unit that would register as a
> SIP extension then would handle auto-answer that I could send a sound file
> to.
>
>
Seems lik
We currently have an Asterisk system that is hooked up to our old paging
speakers via sound card, plugged into two amps.
Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both
warehouses are around 30k square feet. Both have a large number of printing
presses.
The comput
Roy Abshire wrote:
I only have my messages debug file that doesn't show any errors when
placing calls. I thought no one was picking up until they called back
and told me they couldn't hear me. After calling my own phone line...I
picked up and all I hear is it still ringing.
How do I properly deb
Roy Abshire wrote:
Outoing calls I make using Motif Google Voice Calls continue ringing
even after the other end picks up.
I have to restart Asterisk to resolve the issue.
I don't see any errors.
It's not recognizing that the other party picked up the phone and
restarting Asterisk fixes it only
Outoing calls I make using Motif Google Voice Calls continue ringing
even after the other end picks up.
I have to restart Asterisk to resolve the issue.
I don't see any errors.
It's not recognizing that the other party picked up the phone and
restarting Asterisk fixes it only for a day.
--
C
Edwin wrote:
> i recently setup an Asterisk system in Hong Kong. their phone
> company told me that their T1 PRI switch type is Primary-NTT.
> however in chan_dahdi.conf there's no such option. i have it
> set to national. it worked fine for a while, but now suddenly
> stop working. in coming call
The main # was forwarded by client for some other # in the past. That # was
not in issue.
So, when we dial the #, Carrier ring once in Trixbox end and route to other
#. It was giving busy tone.
Customer plugged line into analog phone and figured out the issue. once
forwarding disabled, the system
Hello everyone.
I am facing a problem with Asterisk version 1.6.2.24. What happens is that
when you receive a call (A) in the queue and a member (B) answers the call
normally. At the time that "A" or "B" off the member "B" of the queue
continues as "busy".
The problem started occurring when my pr
HI Andrew,
Show your queuecontrol context. You should have extension s with priority
1 in this context.
--Satish Barot
On Mon, Jan 7, 2013 at 12:08 PM, Andrew White wrote:
> Hi Satish,
>
> ** **
>
> Thanks for your response – sorry on the slow reply.
>
> ** **
>
> So I’ve tried the fo
On Thu, Jan 03, 2013 at 09:44:43AM +, A J Stiles wrote:
> On Thursday 03 January 2013, Selva M wrote:
> > Hi,
> >
> > I setup PBX with A400P 4 x FXo board. There are one analog line plugged
> > into port 1.
> >
> > Internal extension cane make calls to PSTN without any issue.
> >
> > When
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