Hi,
1. From chan_dahdi.conf.sample (asterisk 1.6.1.18) :
; Switchtype: Only used for PRI.
;
; national: National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess: ATT 4ESS
; 5ess: Lucent 5ESS
; euroisdn: EuroISDN (common in Europe)
; ni1:Old
We got it fixed.
In reality, the output of lsdahdi did not look suspicious.
Following a suggestion of Emanuele Carbone I looked at the output of
/etc/init.d/dahdi start and realized that the driver was not loading because
because we had changed the config to use oslec (sw echo cancellation)
Hi,
1. Has someone met any success at all, connecting a Digium B410P to a Patton
Smartnode 4638 (with latest 5.3 firmware) ?
2. If positive, then, which signalling was used on both sides ?
My project's goal is use a Patton Smartnode 4638 to act as telco BRI lines,
from a B410P-enabled asterisk
On Tue, May 04, 2010 at 06:46:59PM +0200, isca...@free.fr wrote:
- Create a SSH tunnel from the Windows client to the Asterisk server using
putty
(redirecting ports used for VoIP)
= it doesn't work because either SIP/RTP or IAX2 protocol are based on
UDP
so that SSH tunneling isn't
2010/5/9 Tzafrir Cohen tzafrir.co...@xorcom.com
On Sat, May 08, 2010 at 08:02:49PM +0200, Olivier wrote:
2010/5/8 Olivier oza_4...@yahoo.fr
Hi,
I'm trying to dial from one Asterisk box to a Patton 4638 BRI gateway.
I'm only getting this :
[May 8 15:08:24] WARNING[16797]:
Hey Guys
We are replacing a BM4 with a trixbox (asterisk) virtual numbers as the
customer wants to move the callcentre.
They are asking for an equiv to the ipview
I gather HUD may be or the panel view
The problem is that we need to see
(a) total calls in the queue
(b) calls for specific DID
On Sun, 2010-05-09 at 13:34 +0300, Tzafrir Cohen wrote:
On Tue, May 04, 2010 at 06:46:59PM +0200, isca...@free.fr wrote:
- Create a SSH tunnel from the Windows client to the Asterisk server using
putty
(redirecting ports used for VoIP)
= it doesn't work because either SIP/RTP or
Dear all,
We have been using asterisk for 4 years. Now we have got problems which
occurs during the attended transfer, blind transfer and answering
machine(during the pressing of 999).
During attended transfer and blind transfer , sometimes we cannot hear
the sound of 'pbx-transfer'. The same
Dear all,
We have been using asterisk for 4 years. Now we have got problems which
occurs during the attended transfer.
During attended transfer, sometimes we cannot hear the sound of
'pbx-transfer'. Sometimes we can hear little portion of
'pbx-transfer's sound. That means sound also become noisy.
Process of elemination. Test with multiple phones, check the codec being used
and make sure the file is there and available.
- Original Message -
From: kamrun nahar bina
To: asterisk-users@lists.digium.com
Sent: Friday, May 07, 2010 07:33
Subject: [asterisk-users] Problem of
- Original Message -
From: Tilghman Lesher tles...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 07, 2010 22:39
Subject: Re: [asterisk-users] Getting presence working in 1.6.2
On Friday 07 May 2010 13:59:23
On Wed, May 5, 2010 at 7:10 PM, Adrian Marsh adrian.ma...@ubiquisys.comwrote:
Anyone have any experience with a Japanese local VoIP termination
supplier?
I’ve emailed a few companies looking to setup some PSTN to SIP and SIP to
PSTN termination, but no luck so far.
Thanks,
Adrian
Dear,
We have tested in several phone like snom 300, LINKSYS, s-lite. The same
error occurs in every phone. Our codec is ulaw. and we have checked that the
file is there and available.
Is there any other solution? or is it the problem of db load, is it
possible?
Thanks in advance.
nahar
On Mon,
its possible
ask the same question trixbox forum
Ram
On Sun, May 9, 2010 at 6:45 PM, Samantha saman...@femtech.com.au wrote:
Hey Guys
We are replacing a BM4 with a trixbox (asterisk) virtual numbers as the
customer wants to move the callcentre.
They are asking for an equiv to the ipview
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