On Fri, 2010-01-29 at 07:29 -0600, Kevin P. Fleming wrote:
Szasz Szabolcs wrote:
How can I disable comfort noise on Asterisk?
Asterisk does not have a comfort noise generator, so there is nothing to
disable. You'll have to be more specific about what you are trying to
accomplish.
--
On Thu, 2010-01-28 at 23:11 -0600, Karl Fife wrote:
Appears completely resolved!
No more home-spun patches!
Thanks!
-K
It's *not* fixed here:
DAHDI Version: 2.2.1 Echo Canceller: MG2
But as is depressingly the 'norm' for Asterisk it comes back to bitching
about hardware 'buy an expensive
On Wed, 2010-01-13 at 11:13 -0500, Kristian Kielhofner wrote:
On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
..snip..
I've not been able to get that out of them, but I don't *think* It's
Asterisk based because they say:
Unfortunately, our
On Tue, 2010-01-19 at 13:15 +0100, joern wrote:
listu...@spamomania.co.uk wrote:
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to
recognize digits pressed on a keypad coming in from a Sipgate trunk.
There answer was to set this:
dtmfmode=rfc2833
On Fri, 2010-01-15 at 22:26 +, Gordon Henderson wrote:
On Sat, 16 Jan 2010, Tzafrir Cohen wrote:
On Fri, Jan 15, 2010 at 04:06:54PM +, Gordon Henderson wrote:
Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on
1.4.. Nothing special about the hardware -
On Wed, 2010-01-13 at 10:12 -0500, Kristian Kielhofner wrote:
On Wed, Jan 13, 2010 at 2:43 AM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Thanks for that. Looking at the RTP packets I can see two types as you
point out. The first appears to be the audio:
Real-Time
On Wed, 2010-01-13 at 11:13 -0500, Kristian Kielhofner wrote:
On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
..snip..
I've not been able to get that out of them, but I don't *think* It's
Asterisk based because they say:
Unfortunately, our
On Tue, 2010-01-12 at 10:37 -0500, Kristian Kielhofner wrote:
On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net
wrote:
Codec? I've had 2833 do funny things with anything other than a/ulaw
(might just be me though..)
S
--
Codecs other than G711u/a don't
On Tue, 2010-01-12 at 16:52 -0500, Kristian Kielhofner wrote:
On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to
recognize digits pressed on a keypad coming in from a Sipgate trunk.
There answer was to set this:
dtmfmode=rfc2833
in the general section of sip.conf
This has made no difference. I've tried a range of settings (auto,
On Mon, 2010-01-11 at 16:52 +, Steve Howes wrote:
On 11 Jan 2010, at 16:26, listu...@spamomania.co.uk wrote:
This has made no difference. I've tried a range of settings (auto,
rfc2833,info) but no matter what, it plain refuses to pick up key
presses.
snip
It's extremely frustrating
On Sun, 2010-01-10 at 00:25 -0800, Nitin Bahadur wrote:
Hi Tzafrir,
Some more background...I have a comcast phone line
which I have connected to my FXO port. When I call my
number, it goes directly to comcast voicemailin other words,
there is no ringing tone and pickup by asterisk.
I'm struggle to answer a simple question. One user at extension 4000
wants a custom .gsm file to play for their mailbox. I can't figure where
to put it/what to set in voicemail.conf to achieve this:
voicemail.conf
4000 = 4000,system,voicem...@net
Relevant extensions.conf line:
exten =
On Wed, 2009-12-30 at 10:00 -0500, Doug Lytle wrote:
listu...@spamomania.co.uk wrote:
I'm struggle to answer a simple question. One user at extension 4000
wants a custom .gsm file to play for their mailbox. I can't figure where
to put it/what to set in voicemail.conf to achieve
On Wed, 2009-12-30 at 15:18 +, Ishfaq Malik wrote:
Get the customer to log into their voicemail mailbox and follow the
instructions to record an unavailable message (Options 0 then 1 if there
are no messages I think)
Then in the conf you need
exten = 2,n,VoiceMail(4...@voicemail,u)
On Mon, 2009-12-28 at 12:11 -0600, James A. Shigley wrote:
Alright I have a SIP phone located off premises with a very annoying
issue.
Well I say a sip phone it is actually two phones hooked to a Cisco Spa
2102
Link: http://www.cisco.com/en/US/products/ps10026/index.html
Looks
The answer is probably no, but I have a bypass PSTN phone ahead of my
Asterisk 1.6 box.
I noticed when a call is answered on this bypass phone, the 'record'
option still partially operates on Asterisk, but stops after the ring
detection goes low.
Is it possible to have Asterisk record when a
On Thu, 2009-12-24 at 18:53 +0100, Gergo Csibra wrote:
Thursday, December 24, 2009, 5:41:46 PM, Danny wrote:
Just my opinion; unless you are recording long or many long calls, you
should record to your local drive, then copy the files to the USB drive.
Asterisk is a very good tool - you
Dave Wrote:
It looks like whatever is being transmitted, or the response, isn't
getting through. Possibly due to NAT or a firewall? It would help if you
described the scenario where this is occurring.
Indeed, my post was gibberish :-O
This was a 'nat' issue, but not in the traditional sense.
Hi,
How would I go about troubleshooting this:
[Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum
retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for
seqno 101 (Critical Response) -- See doc/sip-retransmit.txt.
[Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397
On Sun, 2009-12-20 at 16:16 -0400, Tim Johnson wrote:
Possibly OT?
I've hooked up a Linksys PAP2 to my Asterisk 1.6 fairly painlessly. The
only issue I can't beat with it is the dial delay when calling internal
or external numbers.
No matter what it seems to take 10 -15 seconds to
Possibly OT?
I've hooked up a Linksys PAP2 to my Asterisk 1.6 fairly painlessly. The
only issue I can't beat with it is the dial delay when calling internal
or external numbers.
No matter what it seems to take 10 -15 seconds to actually dial. I've
altered the device removing all *xx combos and
Some recent issues I had with hardware seem to come back to not
understanding two very similarly named files:
/etc/asterisk/dahdi-channels.conf
/etc/asterisk/chan_dahdi.conf
I've modified the chan_dahdi.conf to work now, but it would appear all I
needed to do was include dahdi-channels.conf in
I've spent a week playing with Asterisk 1.6 and I love it. What a
brilliant piece of software!
Progress and learning have been reasonably good. I have external SIP
provider calls coming in and have put together a little call platform
and I'm stunned at the flexibility.
There is one issue for me.
On Mon, 2009-12-14 at 14:17 -0300, VinÃcius Fontes wrote:
I have never used that card myself, but I have never seen an analog board
reporting a RED alarm. Probably there is something incorrect in your
configuration. Please post your /etc/dahdi/system.conf and
/etc/asterisk/chan_dahdi.conf.
On Mon, 2009-12-14 at 11:11 -0600, Tilghman Lesher wrote:
On Monday 14 December 2009 10:28:08 am listu...@spamomania.co.uk wrote:
I've spent a week playing with Asterisk 1.6 and I love it. What a
brilliant piece of software!
Progress and learning have been reasonably good. I have external
On Mon, 2009-12-14 at 14:10 -0600, Tilghman Lesher wrote:
I don't want to start a war, but there is a square to that. I'm new to
Asterisk having spent years in analogue telephony. If I can get a test
Asterisk working on a cheap clone card without a hitch, I'm most likely
to expand this
On Mon, 2009-12-14 at 22:27 +0200, Tzafrir Cohen wrote:
On Mon, Dec 14, 2009 at 02:17:39PM -0300, VinÃcius Fontes wrote:
I have never used that card myself, but I have never seen an analog
board reporting a RED alarm.
Ahem. Wcfxo always has (AFAIR). Red alarm means that no line is
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