OK, after my last message about fax detection, I feel a bit better informed and
able to press forward. I started looking into this because I was getting lots
of false positive fax detection errors in the logs with faxdetect=both set in
chan_dahdi.conf.
Anyhow, I do not currently use fax
Hi all,
I have a system installation in Guam with two trunks. One has a DSL service
riding on it with the usual filter. That channel however keeps throwing
alarms. I bypassed the filter and it stopped throwing alarms, but of course
the high frequencies annoy the users. I swapped the filters and
I'm pretty impressed by how well (comparatively) google voice does in
doing voice mail transcripts. So I'd like to have google do my local
voice mail, and then email the transcript.
So I set up extensions.conf:
exten =s,n,Dial(${House_Phones},36) ; this should be six rings
exten
If you do get a Polycom, the old 501 (discontinued) have a louder ring
(or can be configured to have a louder ring, don`t quite remember)
then the newer ones. But the others are right: it's not meant for
this, at least not in a noisy environment. What can work though is a
Polycom 321,
Hello list,
I'm having DTMF-troubles with a Snom phone. I want to know if it's the
Snom or Asterisk that makes the trouble.
I'm playing a prompt, then make a choice for 2 :
[Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] --
SIP/test1-0701 Playing
Have recently installed some Snom phones into an office. Phones are
natted and connect to a 1.4 server on a public IP
We can make outgoing calls, but are unable to answer incoming calls.
The phone rings, but the call cannot be picked up. Other phones on
other sites connected to the server are
On Tue, 04 Jan 2011 17:57:27 +, Sebastian s...@open-t.co.uk
wrote:
Sorry to keep on butting in. I've been interested in SIP on Android for
a while now - so this just gave me more incentives to actually do the
research :-)
No problem. I hadn't thought about using a 3G connection to register
On Wed, 05 Jan 2011 11:49:40 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
As I told, the best SIP client I had is Nokias one. Fully integrated,
working out of the box.
Thanks much for the feedback. I was mentioning OpenVPN because I
assumed 3G carriers blocked SIP, but your experience
On Wed, 2011-01-05 at 15:47 +, Ishfaq Malik wrote:
Hi
We've been running asterisk 1.4.17 (deb package) in a production
environment for some while now and are finally taken the plunge to
update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
Architecture
I have upgraded
On 01/05/2011 08:12 PM, Thomas Rymes wrote:
OK, after my last message about fax detection, I feel a bit better informed and
able to press forward. I started looking into this because I was getting lots
of false positive fax detection errors in the logs with faxdetect=both set in
Does Asterisk, currently using version 1.4, get any more information about the
result of an outbound call made over a PRI line compared to a call via a SIP
trunk?
As an example, in a PRI call there is this message that shows up on the console:
[2011-01-05 14:59:02] -- Channel 23 detected a
On 01/05/2011 01:51 PM, Tom Rymes wrote:
On 01/05/2011 7:50 AM, Andy Graybeal wrote:
We've got two noisy kitchens that need to talk back and forth.
Andy,
Why, exactly, are you trying to combine an inter-kitchen intercom and
your phone system? Might it make more sense to have a
On Wed, Jan 5, 2011 at 6:59 PM, Myles Wakeham my...@techsol.org wrote:
For some reason our Asterisk box is doing something really unusual following
applying a routine update to CentOS 5 on Monday.
We have Asterisk 1.4.2 and its been working great for years. But now when
the phone system
On 01/06/2011 05:25 AM, Tim Panton wrote:
On 5 Jan 2011, at 13:07, Steve Underwood wrote:
G.722.1 is a 7kHz bandwidth codec. G.722.1C is a stretched version offering
14kHz bandwidth. These are most often found in Polycom phones, but they are
available elsewhere. The only widely supported HD
We should also be very clear that the Siren codecs are supported on the
Polycom SoundStation conference phones and the VVX-1500 Business Media
Phones. These codecs are not supported in the SoundPoint desk phones.
The SoundPoint series support the more basic G.722 codec in the
On Wed, 5 Jan 2011, James Lamanna wrote:
See the following SIP trace.
Where in the world does Asterisk get port 1025 to respond to?
This is asterisk 1.6.x.
Hi James,
I'm sure it would be the NAT translated port on the public side of the
customer's firewall...
j
Thanks.
-- James
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