I'm trying to configure my Asterisk machine to work with Vitelity's
vMobile service. I can place calls to the vMobile device and it rings
as expected. However, I have no audio in either direction. There's
no NAT involved though. My asterisk machine has a public IP address
with port 5060 and
I'm using chan_motif with Asterisk 11. It still works. I actually
received an email from google yesterday that there had been no traffic on
my number lately so the number would be reclaimed. I had switched my
outgoing away from GV several months ago when they were supposed to
discontinue the
ulaw
On Wed, Jul 10, 2013 at 7:40 AM, basteon bast...@gmail.com wrote:
Hi,
What codec do you use with yours subscribers?
On 9 July 2013 23:45, Chris Gentle gent...@gmail.com wrote:
Is there any way I can improve the audio quality in a confbridge in
Asterisk 11? I've changed
On Wed, Jul 10, 2013 at 9:16 AM, Matthew J. Roth mr...@imminc.com wrote:
The sampling frequency for u-law is 8,000 Hz. You can't produce a recording
with higher quality than the source, so you'd have to switch to a wideband
codec
to improve the conferences and recordings [1] [2].
OK, thanks
OK, thanks for the advice. No, there's no filter so I'll look into that.
On Wed, Jul 10, 2013 at 3:02 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 07/10/2013 06:46 PM, Chris Gentle wrote:
[snip]
and then others can connect via SIP. For some reason, when the
speaker says
Is there any way I can improve the audio quality in a confbridge in
Asterisk 11? I've changed the internal_sample_rate setting to 44100
but that doesn't seem to make any difference. I would also think this
would make my confbridge recordings 44100 but they all end up as 8000.
Am I completely
Google the number and you can probably find other complaints and
possibly who it is. Not that it will matter, there's nothing you can
do but block it.
My approach to call filtering is:
Deny All
Allow Some
I have a whitelist of callers I always want to accept that may include
businesses outside
.
Thanks,
--
Joseph
On 06/13/13 14:30, Chris Gentle wrote:
Google the number and you can probably find other complaints and
possibly who it is. Not that it will matter, there's nothing you can
do but block it.
My approach to call filtering is:
Deny All
Allow Some
I have a whitelist
I have a confbridge setup that feeds the conference from the ALSA
microphone input (this is the conference leader) and then uses
app_ices to send the conference audio to icecast.
I start the conference leader like this:
console dial 1000_admin@conferences
I join the ices user to the confbridge
On Mon, Jun 3, 2013 at 11:52 AM, Matthew Jordan mjor...@digium.com wrote:
(1) Verify that with all 'normal' channel drivers, such as chan_sip,
that the Conference tears down correctly.
OK, looks like this is the problem. Taking chan_local out of the
picture, I tested it with an incoming SIP
OK, I need a bit of help here. I'm configuring a new Asterisk 11
system and I accidentally let my firewall rules drop for a day or so.
When I logged in today, I found messages like the ones below on my
asterisk console. Obviously somebody was trying to take advantage of
my carelessness. So can
on...
you could enable sip tracing to get more information.
maybe you should change the 'allowguest' option in sip.conf..?
regards,
yves
Am 31.05.2013 23:57, schrieb Chris Gentle:
OK, I need a bit of help here. I'm configuring a new Asterisk 11
system and I accidentally let my firewall rules
kind of interrupt hammering going on here with
my particular hardware. Even before the audio completely fell apart I
could hear some little pops that sounded like the interrupts were not
being serviced fast enough.
On Mon, May 6, 2013 at 8:31 PM, Chris Gentle gent...@gmail.com wrote:
OK
OK, somebody may have a much better way of doing what I'm attempting. If
so, I'm open to suggestions.
I am trying to configure confbridge to create a conference room with an
audio stream coming from my sound card. The idea is for a group of people
to be able to call in and listen to someone
On Sat, Mar 23, 2013 at 10:45 AM, Harley Peters
har...@thepetersclan.com wrote:
I had it set to 1 originally and it worked fine at first then suddenly
stopped.
It drove me crazy until I ran across this link:
http://iprouteth0.blogspot.com/2013/01/new-thoughts-troubleshooting-google.html
Set
I just wanted to send out some information that will hopefully help
others. I don't know, maybe I'm the only one that's been having
problems with this. I've been pulling my hair out for a while
wondering why Google would not send my incoming calls to my Asterisk
box. The calls would just roll
I'm currently running Asterisk 11.2.1 and I've noticed that when asterisk
has been up for a while (usually about a day), outgoing calls through
GoogleVoice fail to complete. I hear it ringing on my end but the caller
never hears the phone ring. A simple restart of Asterisk seems to clear it
up
Awesome, thanks. I'll give it a try.
On Mar 11, 2013 4:56 PM, Joshua Colp jc...@digium.com wrote:
Chris Gentle wrote:
I'm currently running Asterisk 11.2.1 and I've noticed that when
asterisk has been up for a while (usually about a day), outgoing calls
through GoogleVoice fail to complete
On Wed, Jan 2, 2013 at 9:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote:
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
apt-get install asterisk
Does anyone know of any
On Wed, Jan 2, 2013 at 10:19 AM, Dan Jenkins
dan.jenk...@holidayextras.comwrote:
On 2 January 2013 16:16, Chris Gentle gent...@gmail.com wrote:
Does anyone know of any asterisk 11 packages for the Pi? I ended up
compiling it myself this weekend. Took a while.
Take a look at http
I need some advice on how to implement something in my dialplan.
Here's the scenario. A call comes in on my [incoming] context and I answer
it. The call turns out to be for my wife and she needs to answer it on a
different
handset somewhere else in the house.
I've tried call parking but the
On Mon, Nov 19, 2012 at 6:23 PM, Jared Baxley jared.bax...@gmail.comwrote:
You can park the call, set the timeout low, and have it return to a ring
group.
Thanks to everyone for the suggestions. I decided to try this approach
first and I think I have it working. However, I found a slight
On Wed, Jun 13, 2012 at 5:12 AM, Administrator TOOTAI ad...@tootai.netwrote:
someone knows when asterisk binary packages will be available on
asterisk.org for Ubuntu precise (aka 12.04)?
I did a fresh install of Ubuntu Server 12.04 LTS (precise) on my system
today and was a bit surprised to
I have two GV numbers. Both are configured to send calls to my Asterisk
1.8.13.0 box using the Google chat interface. At one time I had both
working with Asterisk. Now, for whatever reason, one of them has stopped
sending incoming calls to my asterisk box and instead just rolls to GV
voicemail.
On Wed, Jun 20, 2012 at 12:14 PM, Warren Selby wcse...@selbytech.comwrote:
As you said, GV and asterisk integration is unstable at best. I haven't
worked with it in a while, to be honest. But, with all that being said,
I'm not opposed to popping my GV test box back online and helping to
I have a TDM410 with one FXO and one FXS. I've been running dahdi 2.5.0.2
without any problems. A couple of weeks ago I upgraded to 2.6.0 and found
that caller ID was no long working for me. All calls came in with a blank
caller id. I reverted back to 2.5.0.2 and everything was happy again.
On Thu, Mar 15, 2012 at 10:08 AM, Shaun Ruffell sruff...@digium.com wrote:
Hi Chris,
I believe this is fixed in the head of the 2.6 branch. We're
prepping a 2.6.0.1 release now...
Hey Shaun. Thanks for the quick reply. I applied the patch for the bug to
my 2.6.0 and it works fine. I've
On Sat, Jun 4, 2011 at 1:05 PM, sean darcy seandar...@gmail.com wrote:
I'm trying to set up csipsimple on my Droid X. But no joy. Can't get it to
register.
It works for me on my HTC Thunderbolt. I can get it to register both when
inside my local network and also when I'm outside my network
On Mon, Apr 11, 2011 at 8:47 AM, Brian Henning bhenn...@pineinst.comwrote:
H. I do see this in the /var/log/asterisk/messages log:
[Apr 5 00:05:36] NOTICE[9579] chan_dahdi.c: Got event 4 (Alarm)...
[Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 5 (No more alarm)...
[Apr 5
On Mon, Apr 4, 2011 at 10:25 AM, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
No problem. You just specify accountn...@gmail.com.
exten = accountn...@gmail.com,1,Answer()
exten = accountn...@gmail.com,n,Wait(2)
exten = accountn...@gmail.com,n,SendDTMF(1)
exten =
Hello. I would like to configure Asterisk to accept incoming calls from two
different GoogleVoice numbers via gtalk and jabber. I'm running Asterisk
1.8.3.2 and I can get one number working just fine. However, I can't figure
out how to modify the gtalk.conf file shown on the Asterisk wiki site
Anybody else noticed that caller id for outbound calls via Google Voice
seems to be broken? It seems to be a Google Voice problem though, not an
asterisk issue.
--
Chris
--
_
-- Bandwidth and Colocation Provided by
On Thu, Feb 24, 2011 at 9:08 AM, William Stillwell
will...@stillwellsoft.com wrote:
Yes.. google it
I did. :)
This is what I have done to resolve it (I posted a few days ago on this)
exten = _9NXXNXX,1,Dial(gtalk/(value in gtalk.conf)/+1(googlevoice#)@
I've been playing with call parking in Asterisk 1.8.1. I'm able to park a
call and then pick it back up. However, on the second attempt, the #72 DTMF
is ignored. Asterisk just passes that DTMF on to the caller and the call
parking never happens. Shouldn't I be able to park a call more than
On Mon, Nov 29, 2010 at 11:07 AM, Roger Burton West ro...@firedrake.orgwrote:
The desired result is that user A's phone rings; when he picks it up,
user B is dialled, and user A's channel is connected to that. (This is
to be a back-end for a web-based address book.)
This is click-to-call.
On Tue, Nov 16, 2010 at 8:28 AM, Gilles codecompl...@free.fr wrote:
Hello
For users who 1) don't have a QoS-capable ADSL router and 2) would
like to run Asterisk with a couple of SIP trunks, I was wondering what
hardware is recommend to run any of the main open-source *WRT projects
to which
On Thu, Nov 18, 2010 at 9:20 AM, jon pounder j...@inline.net wrote:
I have a similar setup in an office but sip directly back to the main
server - not sure what the value add to the local asterisk is, except
intercom calls would not have to leave the lan, but isn't that the purpose
of
On Wed, Nov 10, 2010 at 8:52 AM, Paulo Santos paulo.r.san...@sapo.ptwrote:
Hello list,
I'm having some issues with some phones that don't stop ringing after
the call is answered somewhere else.
Basically, a call comes, enters a queue and all the phones in the queue
ring. The problem is
When I call into my Asterisk box via my VoIP line (using gsm codec) and then
try to make an outgoing DISA call over PSTN I get the following:
[Nov 1 15:12:54] WARNING[17694]: chan_dahdi.c:8930 dahdi_write: Cannot
handle frames in gsm format
[Nov 1 15:12:54] WARNING[17694]: app_dial.c:1401
I'm running Asterisk 1.6.2.1 with DAHDI 2.2.1 using a TDM410P. I have
callprogress=yes in chan_dahdi.conf because, from everything I've read, it
is needed when using call files over PSTN, which I DO use occasionally.
I know that callprogress=yes is experimental and causes some issues.
We've
On Thu, Apr 15, 2010 at 4:06 PM, Baji Panchumarti
baji.panchuma...@gmail.com wrote:
Steve, Chris :
I too had this problem and the solution was not tweaking
the AMD parameters, but playing a short audio file (even
a really really short one) before executing the AMD function.
The key
I know that Asterisk can use the system's sound card as the output device
for a console channel. However, I'm using Asterisk call files and would
like to be able to hear the calls over a set of speakers as the call files
are being processed. Basically I'm wanting to listen in on the calls as
On Tue, Mar 23, 2010 at 9:06 PM, Steve Moran s...@matara.net wrote:
I am running Asterisk and using Answer machine detection with call files on
a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD
is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over
After an upgrade to asterisk 1.6.2.1 I'm unable to make outgoing calls via
Vitelity. I get lots of these on my asterisk console:
[Jan 27 08:58:41] WARNING[25653]: chan_sip.c:3581 __sip_xmit: sip_xmit of
0x834ae08 (len 927) to 64.2.142.18:0 returned -1:
Address family not supported by
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