hi everybody,my problem when a person me call of exterieure (network to connect via
vpn) and that it hangs up again her accosting by in timeout during 30
second then an error message appparait("critical transaction
failed:client non-INVITE transaction[Trying]: timed out") during all
this time it th
Hi!I've a problem with asterisk,it doesn't start..it's stopped..I used the amportal start command and this is the result:
SETTING FILE PERMISSIONSPermissions OK
STARTING ASTERISKAsterisk ended with exit status 1Asterisk died with code 1.Automatically restarting Asterisk.Asterisk ended with exit sta
Hi, I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone. Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to im
Hi,
I have a digium T1 card installed on my Asterisk box. Protocol is PRI.
I am trying to setup so that the box can send and receive faxes. Being
able to receive faxes is a lot more important than being able to send.
I tried spandsp-0.0.2pre25, with proper app_rxfax.c file. But I am not
able to r
Hi,
I have seen a few threads where people have applied packetization
patch and have varied the packetizing rate of RTP/SIP and hence
reducing the bandwidth required for the call.
Is there a way to do the same with IAX?.
Will the tos=0x08 (highthroughput), or using the bandwidth directive help
Julio Arruda wrote:
Anyone tried the new PSTN/FXO port in the new SPA 3102 FXO/FXS adapter ?
From a quick test (got mine yesterday), seems like it is not
recognizing Caller ID from PSTN/FXO port..
It's a known bug in the current firmware.
HTH,
Vahan
___
Hi,
I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box with
SIP. Asterisk always returns "Username/Password mismatch".
I've tried all configurations that was on the Quintum's manual, but no
success.
I've tested the same username and password with a Linksys (PAP2-NA) with
We suffered no outbound downtime that was a 'show-stopper' proof
positive that JJ was and always will be there, however the TF REsporgs
are still killing me. You are working your butt off and I appreciate it,
good luck and hope to hear ringing in my ears soon. :-)
Alex (OpSys)
> -Original M
Tom Vile wrote:
Definately adds stress to your life when a provider disconnects you
but I can see that they are trying to square things away with there
customers and it looks like good changes are in the works.
Finally, after how many months of downtime?
Jeremy McNamara
Andrew Kohlsmith wrote:
On Tuesday 23 May 2006 10:48, Bart Fisher wrote:
Can anyone clue me in about these T400 T1 cards I see advertised? I hear
they are Digium
Clones. Is there some reason to avoid these? How do they compare to
TE410P's for example.
Google for the performance dat
Definately adds stress to your life when a provider disconnects you
but I can see that they are trying to square things away with there
customers and it looks like good changes are in the works.
On 5/23/06, Manny A. Wise <[EMAIL PROTECTED]> wrote:
I never liked Jeremy, having that out of the way
Hi All,
Can someone please advise me about configuring my Polycom IP600? I have an
account with a SIP based IP Centrex provider. The basic SIP info and line 1
config points to them. That's
working fine.
I'd like to register line 2 with my own asterisk server. I've tried putting the
basic regi
On Tue, 2006-05-23 at 10:46 -0700, Gary Richardson wrote:
> Hey guys,
>
> When a call comes in via the PSTN to our Call Manager 3.2 and is
> forwarded (via unity and H323), the caller id is set to our Unity
> Voicemail instead of the caller id from the PSTN. We're using the
> oh323 channel in this
I never liked Jeremy, having that out of the way, :)
What happen to him can happen to ANYONE!
It happened to Broadvoice big time Also Vonage!!.. but they are more
prepared to deal with the root cause!! They have more resources!! And more
MONEY!!! It has nothing to do with reputation!!!
Don't
Well, is very, very sad to see that
every time, we start saying who is the best and more reliable, that company automatically
start going down hill….. I used to love T**iax…. But lately..they
are not the same as last year…I can’t call them “reliable”
as they used to be…. I used to say that
I think the point everyone is making is that no reputable company
would have had this happen. Can you see Vonage losing all their DIDs?
No! NuFone clearly did something that screwed their contract with
their CLEC...
On 5/23/06, Jens Vagelpohl <[EMAIL PROTECTED]> wrote:
-BEGIN PGP SIGNED
Álvaro Palma wrote:
> [test_context]
> exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag
This is wrong; multiple options to Dial go into the same argument
position, not separated.
exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2)j)
should do what you want.
___
Anyone tried the new PSTN/FXO port in the new SPA 3102 FXO/FXS adapter ?
From a quick test (got mine yesterday), seems like it is not
recognizing Caller ID from PSTN/FXO port..
Using the same configuration as a Sipura 3000 (to be sent to
mother-in-law POP :-), no Caller ID at all, (I've even
Thanks Mike,
This is exactly what I wanted :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dr.
Michael J. Chudobiak
Sent: Monday, 22 May 2006 11:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Snom 320 Sh
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On 23 May 2006, at 23:55, Alexander Lopez wrote:
Are the 800 numbers you have new (post-outage) of existing
(pre-outage)??
SNIP
It is working, I have a 800-number with them.
I have a new one right now, but have initiated the porting process to
Are the 800 numbers you have new (post-outage) of existing
(pre-outage)??
SNIP
>
> It is working, I have a 800-number with them.
>
> jens
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Michael Knill wrote:
I have not installed mpg123 so it must be native.
I've had issues with Native MOH over IAX2 while parking. Very distorted
MOH and high pitch squealing. But, Native MOH from a PSTN or just plain
putting a call on hold via the SIP phones is fine. I moved the Parking
Lo
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Here's one for all the naysayers: I only sent an email to NuFone
accounting to inquire about that $2.50/month fee and they're falling
over themselves to not only get all my questions answered but to also
helping me getting my account set up in t
From:
Jon Scottorn <[EMAIL PROTECTED]>
Date:
Tue, 23 May 2006 12:52:02 -0600
To:
Asterisk Users Mailing List - Non-Commercial Discussion
On Tue, 2006-05-23 at 19:44 +0100, Thomas Kenyon wrote:
Jon Scottorn wrote:
> Hi All,
>
>I have been attempting to get an AGI LCRdialout script to wo
On 5/23/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote:
Bruno de Assumpção Loureiro wrote:
> Hi all,
>
> How to integrate with Oracle database. I think it's possible with AGI,
> it isn't?
>
> Regards,
>
You will probably need to go down the odbc route for oracle.
Read the details on cdr_odbc.conf,
Hi!
I have a question about mixing an IVR and agents in such a way that a
caller can listen to information from IVR (PerlAGI) while agents are
busy. As soon as agent is available the channel whit IVR should be
notified to ring the operator or both should go to MeetMe room. How can
one channel
Hello all!
Can anyone think of an *easy* way to get the IP number of the server
running asterisk from within the dialplan?
Thank you in advance!
Steve
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I have a weird situation. A polycom phone is configured to use system pbx1 as
the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three
systems have identical sip.conf files. The phone is registered on pbx1.
I shut down the Asterisk application on pbx1. I make a call. The phone
On Tuesday 23 May 2006 16:28, Andrew Kohlsmith wrote:
> On Tuesday 23 May 2006 14:54, Cosmin Prund wrote:
> > Is there a known "hack" or patch to blow Asterisk's echo canceler up to
> > 128ms? Or at least 64 ms?
>
> It's real easy... at least for us code tinkerers.
It's even easier now, as Digium
That's wrong. It compiles, it can be loaded. But it does nothing
if merged inside a 1.2.7.x
Perhaps someone found out what is needed to get it operative...
On Tue, May 23, 2006 at 07:57:42PM +0100, Tim Panton wrote:
>
> On 23 May 2006, at 15:19, Carlos Alperin wrote:
>
> > I know that I asked t
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I see it now on the FAQ, but this must be a new thing. I paid $50 in
December 2004 and still have over $39 (yes, I don't use it often). If
I remember correctly the 800 DIDs were advertised as free of monthly
fees, call fees only.
jens
On 23
Hi folks,
I've posted uLaw, aLaw, G729 and G723 variants of the Alison Keenan
British English files.
http://www.enicomms.com/cutglassivr/
Thanks
--
Mark Phillips <[EMAIL PROTECTED]>
___
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Aster
On Tuesday 23 May 2006 14:54, Cosmin Prund wrote:
> Is there a known "hack" or patch to blow Asterisk's echo canceler up to
> 128ms? Or at least 64 ms?
It's real easy... at least for us code tinkerers.
In the zaptel source tree, in zaptel.c, you will see a call to
echo_can_create() with a check
I'm trying to set a dialing rule in my dialplan. As a part of it, from
my point of view, this works wrong
priorityjumping=no
[test_context]
exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag
exten => 1234,2,Playback(digits/2)
exten => 1234,3,Playback(digits/3)
exten => 1
card, IMO. These artifacts are mitigated through the black art and dumb luck
of different chassis, local RF interference
This has me curious.would the RF interference you're thinking
about be interference that affects my SIP phone (earpiece, mouthpiece,
etc), or interference that is affectin
QUEUE_WAITING_COUNT()
in /trunk
On 5/23/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
Is there an Asterisk Application/Function/Variable that returns the current
number of callers in a given queue?
Thanks,
Doug
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$2.50 p/month for 800 DID.
On 5/23/06, Jens Vagelpohl <[EMAIL PROTECTED]> wrote:
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They bill you for having the 800 number? I thought they only did that
for Michigan DIDs. They only bill my actual call time.
jens
On 23 May 2006, at 16:54, Tom Vile w
> -Original Message-
> From: Douglas Garstang
> Sent: Tuesday, May 23, 2006 12:12 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Queue Count
>
>
> Is there an Asterisk Application/Function/Variable that
> returns the current number of callers in a given
Are you doing something funny with the CID on it's way to the phone?
I've got a somewhat similar problem with an Aastra IP phone (yes, I did
say IP): it would NOT ring if the caller id started with an "#". Maybe
your Aastra PSTN phone got some of the same (buggy?) handling of CID's?
Dan Elder
On Tuesday 23 May 2006 13:30, Derek Lee-Wo wrote:
> I'm actually working in the computer telephony field and have been for
> the last 10 years, but I deal mainly with T1s and trunk adapters on
> RS/6000s. I'm a software person so I don't do a huge amount telephony
> configuration, but I have done
On 23 May 2006, at 15:19, Carlos Alperin wrote:
I know that I asked this before:
I need to add res_snmp to a working 1.2.7.
Can I add it to the res directory, modify their makefile and
compile everything again?
When I tried to install everything from svn, I got messages like my
za
Andrew Kohlsmith wrote:
Regarding echo tail: Asterisk's default software echo cancellation is only
32ms at maximum. You can blow this number up to 128ms (where the hardware
echo cancellers sit at), but it costs more in CPU time and memory, and I
think that Digium left it at 32ms as a compromis
On Tue, 2006-05-23 at 19:44 +0100, Thomas Kenyon wrote:
Jon Scottorn wrote:
> Hi All,
>
>I have been attempting to get an AGI LCRdialout script to work.
> Basically what I need to have happen is when someone dials out a
> number the script check to see if it is local if so, go out the ZA
I'm sure it could be but taking the agi route I think would be simpler.
I thought I should be able to get the variable DIALSTATUS from Asterisk. Is this not true.
I have tried these different ways to get it but none have worked yet.
$AGI->get_variable(DIALSTATUS);
$AGI->get_variable(DIALSTA
Bruno de Assumpção Loureiro wrote:
> Hi all,
>
> How to integrate with Oracle database. I think it's possible with AGI,
> it isn't?
>
> Regards,
>
You will probably need to go down the odbc route for oracle.
Read the details on cdr_odbc.conf, extconfig.conf, res_config_odbc.conf
and res_odbc.conf
Jon Scottorn wrote:
> Hi All,
>
>I have been attempting to get an AGI LCRdialout script to work.
> Basically what I need to have happen is when someone dials out a
> number the script check to see if it is local if so, go out the ZAP
> channel. If the ZAP channel is busy, go out the IAX channe
I would have it invoke an AGI script.
[incoming_extensions]
exten => _X.,1,AGI(ManagerControl)
You could have the AGI script have it then jump out to some other
context,extension, or priority in the dialplan or have it handle the call itself.
---johann
Álvaro Palma wrote:
I'm developing an a
Matt,
Matt wrote:
Hi,
If an agent doesn't take a call.. is there some way I can PAUSE them
instead of logging them out?
Sorta, I haven't had time to test it. But you could do something like the
following:
* Use callback agents and have them log into a separate context
(AgentCallbackL
On Tuesday 23 May 2006 13:08, Bart Fisher wrote:
> I'd love to see. Can you provide me your Google search parameters? I end
> up getting a lot of motorcycle data
The paper seems to have disappeared off the face of the internet. I spent the
last hour or so going over google, archive.org and digi
Personally I have found them to be a little touchy.
I have worked on a few and they needed their own version of the Zaptel
driver rather then the stock one.
The genuine Digium items come working right out of the box for me.
Had some issues with newer machines too, though none with older.
Over
Is there an Asterisk Application/Function/Variable that returns the current
number of callers in a given queue?
Thanks,
Doug
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Jens Vagelpohl wrote:
>
> On 23 May 2006, at 16:35, Andrew Kohlsmith wrote:
>
>>> On Tuesday 23 May 2006 10:48, Carlos Chavez wrote:
Now that Nufone is dead, what are other providers of 800 numbers
that
work with Asterisk?
>>>
>>> That's news to me; I terminate about 5kmin/mon
Hey guys,When a call comes in via the PSTN to our Call Manager 3.2 and is forwarded (via unity and H323), the caller id is set to our Unity Voicemail instead of the caller id from the PSTN. We're using the oh323 channel in this case.
Has anyone experienced this issue before? Any solutions?Thanks.
_
Welcome to computer telephony. :-)
I'm actually working in the computer telephony field and have been for
the last 10 years, but I deal mainly with T1s and trunk adapters on
RS/6000s. I'm a software person so I don't do a huge amount telephony
configuration, but I have done my share over time.
On Tuesday 23 May 2006 12:03, Dr. Michael J. Chudobiak wrote:
> The software approach is great in theory, but the hardware echo canceler
> "just works", without configuring anything - in my experience, anyway.
Well, as I said the hardware echo cans are generally tested to comply with
bellcore sta
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They bill you for having the 800 number? I thought they only did that
for Michigan DIDs. They only bill my actual call time.
jens
On 23 May 2006, at 16:54, Tom Vile wrote:
Then you are a luck one aren't you. Haven't had my 800 number for
over
As announced when the Asterisk project converted to Subversion as our
version control system late last year, it is time to decommission our
CVS servers.
As of some time in the next couple of days, the cvs.digium.com and
related servers will disappear; DNS entries for those names will be
removed. I
this is what i have in my event driven router.
exten => X.,1,Answer()
exten => X.,2,MAGI()
exten => X.,3,Hangup()
look in google for info about MAGI patch
regards
On 5/23/06, Álvaro Palma <[EMAIL PROTECTED]> wrote:
I'm developing an application that monitors the state of the incoming
calls us
Hi All,
I have been attempting to get an AGI LCRdialout script to work. Basically what I need to have happen is when someone dials out a number the script check to see if it is local if so, go out the ZAP channel. If the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go
On 5/23/06, BJ Weschke <[EMAIL PROTECTED]> wrote:
On 5/23/06, Matt <[EMAIL PROTECTED]> wrote:
> Right. I understand PauseQueueMember.. that wasn't my question. My
> question was... if a call rings through a member, and they don't
> answer it. Can I have asterisk pause them, INSTEAD of logging
Hi all,
How to integrate with Oracle database. I think it's possible with AGI, it isn't?
Regards,
--
Bruno de Assumpção Loureiro
msn: [EMAIL PROTECTED]
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take this to Asterisk-BizOn 23/05/06, C F <[EMAIL PROTECTED]> wrote:
Define best.On 5/23/06, Crazy Boy <[EMAIL PROTECTED]> wrote:> Hi Friends,>> Can you please tell me who is the best VoIP Service Provider using Asterisk
> (With trail version for sometime) . Waiting for your quick response. Th
I'd love to see. Can you provide me your Google search parameters? I end up
getting a lot of motorcycle data
Bart
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, May 23, 2006 8:38 AM
Subject: Re: [Asterisk-Users] What about T400 T1 cards?
On T
I use Voicepulse as a VIOP provider, the line comes in via a Sipura
3002 box. That's connected to the Asterisk box via a TDM422B POTS
card.
I'd like to add a virtual phone number to my VOIP service so that I
can direct calls to a home business to a different voice mail than
calls to the home pho
On 23/05/06, Matthew Crocker <[EMAIL PROTECTED]> wrote:
> NPANXX breakout.">With a little help from voip-info.org:NPA-NXX-
Where NPA is the 3-digit Numbering Plan Area (Area Code) and NXX
identifies the central office exchange allocated within the NPAs and
are the consecutive last 4 digit
Derek Lee-Wo wrote:
there that might benefit you in a situation like this. Go look through
your zaptel source
tree for "fxotune" and see if it cant possibly correct some of the
problem you're having.
Thanks for this suggestion. I ran the test and activated the settings
with "fxotune -s"and I'
It appears that the PBX sitting between Asterisk and your provider is
not passing on the calling pres flags.
On 5/23/06, Lee Archer <[EMAIL PROTECTED]> wrote:
I have a problem with BT in the UK. Using setcallerpres I can change
the number shown on the recipents phones to Private or unknown but
no, i think there isn't. But definitely you don't need it. Your incoming calls have to go somewhere, at least in a queue. Your manager app will always be able to redirect or drop the channels as needed.
2006/5/23, Álvaro Palma <[EMAIL PROTECTED]>:
I'm developing an application that monitors the sta
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On 23 May 2006, at 16:35, Andrew Kohlsmith wrote:
On Tuesday 23 May 2006 10:48, Carlos Chavez wrote:
Now that Nufone is dead, what are other providers of 800
numbers that
work with Asterisk?
That's news to me; I terminate about 5kmin/mont
Hi,when I am going to compile the zaptel I receive this messageake -C /lib/modules/2.6.12-12mdk/build SUBDIRS=/usr/src/asterisk/zaptel-1.2.5 XPPMOD= modulesmake[1]: Entering directory `/usr/src/linux-2.6.12-12mdk
' WARNING: Symbol version dump /usr/src/linux-2.6.12-12mdk/Module.symvers i
I have not installed mpg123 so it must be native.
Are you using native MOH or mpg123?
Doug___
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While I agree that the Sangoma cards are good, your statement that "software
echo cancellation doesn't really work" is ... incorrect.
Software echo cancel works very well if it's done correctly, if your audio
levels are where the canceller's "sweet spot" is, and the tail is not longer
than the
I can't believe i didn't see that!
i spent ages staring at those damn logs...
And I spent ages scrolling to the bottom of that email.
Please trim your reply to contain only the relevant parts.
This email alone is longer than the full thread
Thanks
__
On 5/23/06, Matt <[EMAIL PROTECTED]> wrote:
Right. I understand PauseQueueMember.. that wasn't my question. My
question was... if a call rings through a member, and they don't
answer it. Can I have asterisk pause them, INSTEAD of logging them
out of the queue? Or, for that matter.. have aste
First of all, I assume that since you're asking the question, you want to
trunk, or send/receive calls that are on the OTHER SIDE of a proxy from you.
Certainly asterisk, as a PBX, can service local IP phones, and connect to PSTN
lines, without regard to ANY internet connection.
Proxy servers
Then you are a luck one aren't you. Haven't had my 800 number for
over a month now but they still bill you for having the number.
Interesting.
On 5/23/06, Jens Vagelpohl <[EMAIL PROTECTED]> wrote:
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On 23 May 2006, at 15:48, Carlos Chavez wrote:
>
Nufone is not dead.
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
> -Original Message-
> From: Carlos Chavez [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, May 23, 2006 10:49 AM
> To: Asterisk
> Subject: [Asterisk-Users] Now that Nufone
Yeah, this email took way too long to hit the list. ChanSpy is my friend
:)
On Tue, 23 May 2006, Aaron Daniel wrote:
I know you can set up monitoring of queued calls, and I'm pretty sure my
question's been answered before, but has anyone devised of a way to actually
barge into a queue channe
On Tuesday 23 May 2006 11:00, Derek Lee-Wo wrote:
> This is what I'm beginning to suspect. I guess someone else in the
> theard stated my question more accurately and what I really want to
> know is if the degradation is noticeable.
Take the time to set it up and tune it correctly. I use TDM400s
From: "Carlos Chavez" <[EMAIL PROTECTED]>
> Now that Nufone is dead, what are other providers of 800 numbers that
> work with Asterisk?
>
Not entirely dead. Yesterday I received an e-mail requiring me to push some
buttons on their dashboard. I think they are still trying.
I switched to exgn.
Little personal preferance here (and hopefully some payback for some help
you gave me a while back). My experience has been that unless you have echo
cancelation on the hardware, the hardware isnt worth purchasing. This holds
for the TDM and T cards right from Digium. If I am going to do a T1 inst
I'm developing an application that monitors the state of the incoming
calls using Manager events. So, as a part of it, I need to "override"
the control of the extensions by the dialplan itself. The problem is
that, if I don't declare the incoming extension, Asterisk hangs up the
call by default
On Tuesday 23 May 2006 10:48, Bart Fisher wrote:
> Can anyone clue me in about these T400 T1 cards I see advertised? I hear
> they are Digium
> Clones. Is there some reason to avoid these? How do they compare to
> TE410P's for example.
Google for the performance data on the TE410. They have so
Asterisk isn't going to put a hiss into the system. Either the hiss
is there on the PSTN already, or your ATA/phone is inducing the hiss.
On 5/23/06, Derek Lee-Wo <[EMAIL PROTECTED]> wrote:
> there that might benefit you in a situation like this. Go look through
> your zaptel source
> tree for
On Tuesday 23 May 2006 10:48, Carlos Chavez wrote:
> Now that Nufone is dead, what are other providers of 800 numbers that
> work with Asterisk?
That's news to me; I terminate about 5kmin/month through them, except for
about 1 week this month when their carrier dropped them. They are most
They are based on the open source zapata telephony tormenta II cards. They
work but offload quite a bit to the server from what I understand. The new
boards are not open source and take quite a bit of load off of the server.
That is my understanding anyways.
Thanks,
Steve
-Orig
I have a problem with BT in the UK. Using setcallerpres I can change
the number shown on the recipents phones to Private or unknown but no
matter what I set my asterisk cid and callerpres to it still displays
the base number of my PRI ddi range.
Regards
Lee
-Original Message-
From: [EM
I just got a DID from www.plainvoip.com the cost is $2.00 a month and 2c
incoming. They also port TF w/ LOA.
Brian
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: Tuesday, May 23, 2006 10:49 AM
To: Asterisk
Subject: [Asterisk-Users
So I saw mention of a way to allow dialing using SIP URI's on Dave McNett's
site at http://slacker.com/~nugget/projects/asterisk/page7
Wow, awesome, I can call anywhere now. However, I think there is a more
elegant way of figuring out whether or not the local * server should handle
a given domain.
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On 23 May 2006, at 15:48, Carlos Chavez wrote:
Now that Nufone is dead, what are other providers of 800
numbers that
work with Asterisk?
Nufone is not dead, works perfectly fine for me.
jens
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Version: GnuPG
ave
> seen on the list.
>
> "I can provide you with tier 1 termination 6/6. I can blend or
> NPANXX breakout."
>
> "We provide US48 termination, blended rate for 1 MOU and above is
> .008 with 6/6."
>
>
> What is 6/6?
>
Define best.
On 5/23/06, Crazy Boy <[EMAIL PROTECTED]> wrote:
Hi Friends,
Can you please tell me who is the best VoIP Service Provider using Asterisk
(With trail version for sometime) . Waiting for your quick response. Thank
you.
Regards,
Chandra.
_
I know you can set up monitoring of queued calls, and I'm pretty sure my
question's been answered before, but has anyone devised of a way to
actually barge into a queue channel so you can do in place monitoring of
calls?
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
On Tuesday 23 May 2006 08:35, Dr. Michael J. Chudobiak wrote:
> Get an FXO card with hardware echo cancellation. I use the Sangoma
> A20002D (four FXO ports with echo cancellation). It definitely costs
> more, but the hardware echo cancellation makes a huge difference in call
> quality! Software ec
Alejandro Vargas wrote:
> 2006/5/23, Thomas Kenyon <[EMAIL PROTECTED]>:
>> Could you give me an example of the macro you use to convert outgoing
>> faxes from iaxmodem to emails?
>
> With hylafax there are defualt files that works, but y changed some of
> them.
>
> I'm attaching the files /var/spoo
No it's not. There will be artifacts in any TDMXXX TigerJet Digium analog
card, IMO. These artifacts are mitigated through the black art and dumb luck
of different chassis, local RF interference, different handsets, different
Asterisk version, etc. But you will most likely never get the exact same
there that might benefit you in a situation like this. Go look through
your zaptel source
tree for "fxotune" and see if it cant possibly correct some of the
problem you're having.
Thanks for this suggestion. I ran the test and activated the settings
with "fxotune -s"and I'll see how it works.
Can anyone clue me in about these T400 T1 cards I see advertised? I hear
they are Digium
Clones. Is there some reason to avoid these? How do they compare to
TE410P's for example.
Bart
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Now that Nufone is dead, what are other providers of 800 numbers that
work with Asterisk?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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--Bandwidth and Colocation provid
Hi
I dont know if it's the best, but for Portugal and to place calls
throwout Europe, www.startel.pt has a good service.
Regards
Joao
Kerry Garrison wrote:
Depends on your location and your requirements. A generic post like
this generally turns into a flame war. Please be MUCH more specific.
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