Hi,
What is the output of the following command?
1. wanrouter status
2. Zap show status in Asterisk CLI
3. wanrouter hwprobe
4. your extensions.conf file
On Mon, May 17, 2010 at 11:31 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
Running the following with a Sangoma A101D PRI card:
Hi,
Can you share successful experience with a SIP hardphone supporting 100 BLF
?
Which phone would you suggest for that ?
(In case that matters, each BLF is supposed to SUBSCRIBE to and reflect the
state (Idle, Ringing, OnCall) of a local extension.
Regards
--
you can use SNOM VoIP phones
On Tue, May 18, 2010 at 11:58 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
Can you share successful experience with a SIP hardphone supporting 100 BLF
?
Which phone would you suggest for that ?
(In case that matters, each BLF is supposed to SUBSCRIBE to and
hello All,
i have one issue with Asterisk Meetme Application
i am recording through Meetme channels through option *'r'* and format for
recording a file is '*wav*'
lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5.
i have very strange problem of meetme_recording ,
once
Please check WAV headers, what is the sample rate of the file? It
should be 8kHz. Does the WAV sound normal when you decrease sample
rate by hand?
You can just upload one WAV for testing - I'll say what may be wrong with it.
On Tue, May 18, 2010 at 9:52 AM, DHAVAL INDRODIYA
2010/5/18 Gopalakrishnan A.N sai...@gmail.com
you can use SNOM VoIP phones
Have you tried them with 100 BLF ?
For instance, Aastra phones are limited to 50 BLF (though you can have much
more buttons).
On Tue, May 18, 2010 at 11:58 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
Can you share
Hi there,
We used to record all the calls with the Monitor function.
Now, I haveimplemented on-demand recording with automon instead...
Everything is working fine apart from the generated filename, which as per
all docs, should be auto-epoch-caller-calleebut in my case, it is
Hello list,
I read on voip-info.org that Asterisk 1.4 support T38 passthrough.
So I guess this means that I can have a Grandstream HT503 with T38
support and an analogue faxmachine on the other side of my Asterisk and
a T38-account with a ITSP on the other side of my Asterisk machine, right ?!
Hi,
The record is not double faster, it's 50% faster (100 seconds original
record - 66.6 seconds recording). Reducing tempo by 33% without
losing pitch sort of fixes the situation, although adds alot garbage
to sound file (you can do this in Audacity).
Sample rate 8kHz is OK, changing it to 5280
I have been trying to get this working with an HT-502, Asterisk 1.4.31, and
Gafachi but no luck so far.
The VSP should send a re-invite for the T.38 media change on detection of
the fax tone.
I'm using canreinvite=yes on the trunk and canreinvite=no on the HT-502
extension. Also have
On Tue, May 18, 2010 at 6:14 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
I read on voip-info.org that Asterisk 1.4 support T38 passthrough.
That may or may not be true. I do not know.
I do know that I've had much better success with fax in 1.6 than I
ever had in 1.4.
My personal
Has any one used this?
U(x[^arg[^...]]):
x - Name of the subroutine to execute via Gosub
arg - Arguments for the Gosub routine
Execute via Gosub the routine x for the *called* channel before
connecting to the calling channel. Arguments can be specified to
the Gosub
Hi guys,
Is it possible to start playing MusicOnHold to the caller but also continue
with the dialplan in single extension, something like this:
exten = s,1,StartPlayingMoh()
exten = s,n,Wait(10)
exten = s,n,Dial(someone...)
exten = s,n,Wait(10)
exten = s,n,Dial(someone else...)
...
Regards,
The simplest way to do this is this:
Exten = s,1,noop(dial with moh)
Exten = s,n,dial(tech/1,10,m)
Exten = s,n,dial(tech/2,10,m)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Sent: Tuesday, May 18,
It happens even with just a few calls (way less than 30).
I'm trying to see if Asus has something to say about this.
In the meantime I'm using trunk=no and it's working fine.
Thanks
Vieri
--- On Fri, 5/14/10, Zoa zoach...@securax.org wrote:
I think that the clock resets would cause no audio
- Vieri rentor...@yahoo.com wrote:
It happens even with just a few calls (way less than 30).
I'm trying to see if Asus has something to say about this.
In the meantime I'm using trunk=no and it's working fine.
Have you enabled trunktimestamps=yes? If I recall, I was able to overcome
Hi Guys,
I'm having a non-obvious issue, i am using Fax for asterisk to receive
faxes, so when i test using a website that send faxes it's working great:
the fax is received and the fax2mail app is called and i get it in my email
box. but when i try using a regular fax machine everything in logs
Anyone have any luck configuring a SIP trunk on a Taqua to talk to Asterisk?
We were initially set up as a subscriber (access line) but that had some
undesirable side-effects, such as quashing the ANI on outbound calls.
Looks like we're going to have to reconfigure the trunk as a network
Has anyone had good results with an on-line database that returns a LATA
based on NPA NXX?
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
--
_
-- Bandwidth and Colocation Provided
On May 18, 2010, at 1:13 PM, Don Kelly wrote:
Has anyone had good results with an on-line database that returns a LATA
based on NPA NXX?
--Don
Don Kelly
There's an online list that you can convert to a locally stored db.
http://www.nanpa.com/nanp1/allutlzd.zip
---fred
http://www.localcallingguide.com/
will give you lots of info.
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Tuesday, May 18, 2010 12:14 PM
To: asterisk-users@lists.digium.com
Subject:
Hi there,
I am stuck with the location issues. It would be easy if you have DID
for each extension so that outgoing caller id would be DID of the
respective extension and also physical address. Now if you are not
able to get DID's for some reason. I am thinking of some situations
and appreciate
Hi
Indeed, limited to only 50 BLF, thats why operator was placed two aastra
phones and set a ring group for both.
Best Regards
Jose Flores Galicia
floj...@gmail.com
BriefCode Code Based Training
2010/5/18 Olivier oza_4...@yahoo.fr
On Tue, May 18, 2010 at 11:58 AM, Olivier oza_4...@yahoo.fr
I have a customer that is using a quad core xeon server with 4 GIG ram
and Te210P card.
Currently this machine is being used for calling out to their own people
as well other programs being run.
anyway they wish to start using it for a 30 person conference bridge.
I presume this is no issue???
On Tue, 18 May 2010, Jerry Geis wrote:
I have a customer that is using a quad core xeon server with 4 GIG ram
and Te210P card.
[snip]
anyway they wish to start using it for a 30 person conference bridge. I
presume this is no issue??? I am running centos 64 and asterisk 1.4.30
[snip]
I
Dumb question - wouldn't it be easier to monitor a web interface than a
phone with 100 lights?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Flores
Galicia
Sent: Tuesday, May 18, 2010 2:32 PM
To: Asterisk Users
Thanks, but in my particular case I need to do pause between dials (using
Wait() command). How could I implement MoH also when Wait is in progress (in
single extensions that is)? Is this even possible, or is the only way to
encapsulate the logic in one extension and do
Here's one way
Exten = s,1,noop(dial with moh)
Exten = s,n,dial(tech/1,10,m)
Exten = s,n,WaitExten(10,m)
Exten = s,n,dial(tech/2,10,m)
Exten = s,n,WaitExten(10,m)
The waitexten(10,m) plays musiconhold waiting for a 1 digit extension. As
long as there's not one in the context, you're good.
Hi all,
I have configured asterisk and a2billing.for inbound i have also configured
did and its forwarded to sip extensions.
But i want to enable queues with inbound numbers(DID).But i could not find a
way to do this in a2billing.
I want enable that if some did comes to asterisk/a2billing it
Hello Everyone,
I must deploy an asterisk system that can support
at least 500 pstn outbound calls.
It's a challenge as it's the first time i'm gonna build such a large
system.
I want to have your advice on hardware, software and so on . What i have in
my plan is a
Hello
I think you can do this using Local Channel
for example I have do so:
queues.conf
[MyQueue]
musicclass = default
ringinuse = yes
strategy=leastrecent
joinempty = yes
timeout=60
retry=5
weight=0
wrapuptime=1
maxlen = 0
announce-frequency = 10
announce-holdtime = no
periodic-announce =
2010/5/18 Danny Nicholas da...@debsinc.com
Dumb question – wouldn’t it be easier to monitor a web interface than a
phone with 100 lights?
Yes and no : operator already has a Flash Operator Panel on its screen.
Information displayed by FOP is richer (you can see who is talking to who)
but
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