would be willing to help me out to understand what needs doing
i'd be very grateful.
Thanks for listening, and hope to hear from you soon!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
:
On Sun, 20 Oct 2013, Notify Me wrote:
I have a small project request on hand where the clients want their
customers to be able to dial in to conduct business over the phone in a
completely automated manner. From my limited understanding this looks a lot
like a call center where one has
for
quite a while.
If you know of another FREE alternative let me know.
While I agree with what others have said about Skype being evil, you can
find another alternative at http://nerdvittles.com/?p=5671
I do not use Skype but I use some of his other stuff and for the most part
it Just works
On Wed, 2 Jan 2013, Robert Rawlinson wrote:
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
Maybe you will find this interesting:
http://nerdvittles.com/?p=3880
Regards,
--
Tom m...@tdiehl.org Spamtrap address
On Sat, 29 Sep 2012, Wrinkled Cheese wrote:
Hello everyone,
I'm having an issue installing AsteriskNOW 2.0.2 on a Dell server. When I
go to install it, with BIOS legacy mode for partition tables, I get as far
as setting up the partition tables. However, the installer then informs me
that GPT
On Thu, 1 Mar 2012, Ralph Green wrote:
Howdy,
I have tried all of these and a few more. PBXinaFlash gave me the
best results, by far. AsteriskNow produced a basic working system. I
could not get any of the others configured to work at all. I should
tell you my restrictions. I
could use wget to download the rpms
you need into a local directory and then do yum localupdate *.rpm to update
them.
Hope this helps.
Regards,
--
Tom m...@tdiehl.org Spamtrap address
me...@tdiehl.org
this mean that for RHEL-6 and its clones there is going to be an
issue?
Regards,
--
Tom m...@tdiehl.org Spamtrap address
me...@tdiehl.org
--
_
-- Bandwidth and Colocation
Hi!
I saw your profile and would like to get to know you better.
Im looking for open, adventurous people, in my area, but we can start here.
Email me back at maris...@email-chatting.com .
Muah!
Marishka ;-)
___
-- Bandwidth and Colocation Provided
I have already get a register, but I can't make a call.I had to setup a listener in order to get the register, but once the register is set I can't make a call in any way.Any hint with that??Thx in advance.Richard OSS [EMAIL PROTECTED] escribió: I think you have to set where to get the libraries
Can someone tell me if passwords are sent in plain text when using IAX?
I have been told already that SIP automatically encrypts the password?
Anyone know of some good Asterisk security links, docs, articles?
Thanks!
___
--Bandwidth and Colocation
agree. I think my phone
was shipped to me in a funny state causing it not to work right. It's a
winner now.
There are some little things I would wish for, but I'm quite happy with it.
Phil
___
--Bandwidth and Colocation provided by Easynews.com
How is the voice quality?
I've just plugged mine back into the charger after having used it
nearly all day. I didn't have any of the problems you've described.
Sorry you're having such bad luck with it. I'm not certain what the
phones are rated to do, but I probably got better than 3 hours
Is there any way to get debug info on res_odbc? I get
the following but this is the last I ever see of
anything ODBC related. Obviously, my extensions are
not working from the database, but I can connect to
ODBC via isql and run queries just fine.
Jul 17 22:12:13 NOTICE[3923]: res_odbc.c:215
How come an outgoing call using my TDM400P immediately
say the call is answered? I'd like to be able to
detect when the call is actually picked up, is this
possible?
If this is normal with analog cards, does the same
thing happen with T1 cards?
-L
Is is possible to initiate a call that is not bridged
to the current channel?
I'd like to initiate an unbridged DIAL, announce the
party that is calling, and then bridge the two calls
together. Is this possible?
Thanks,
-L
__
Do You Yahoo!?
Tired
I have installed a BN8S0 whith chan_misdn (snapshot
09_05_05) in a SuSE 9.0. I have updated the kernel to
2.6.9 in order to make chan_misdn works. And Asterisk
1.0.7 I use mISDN_for_PBX4Linux_2005_03_06 and
mISDNuser_for_PBX4Linux_2005_01_28
It works great, but today I have been doing a test
with
Ok, I have solved my problems by upgrading my
chan_misdn and downgrading my mISDNuser. Now I have
asterisk working with mISDN support.
My problem now is that no matter what I do always see
the link down.
I've plugged the BN8S0 adapter to get the 8 ports
working. When I plug to the ISDN box
PROTECTED] On
Behalf Of me me
Sent: Thursday, 26 May 2005 9:18 PM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] chan_misdn problem
I've installed asterisk 1.0.7 with linux kernel
2.6.3 (patched for mISDN).
I Compile mISDNuser and loaded de modules (hfcmulti,
mISDNdsp) for my
help me??
Thanks.
__
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Nuevos servicios, más seguridad
http://correo.yahoo.es
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http
If the problem is with libtiff, its a problem with every version i've
tried (3.5.7, 3.6.0, 3.6.1, 3.7.1 and 3.7.2)
On 4/30/05, Steve Underwood [EMAIL PROTECTED] wrote:
Me wrote:
Hi all,
I'm trying to use spandsp and asterisk to send faxes. To do so I am
creating tiffs with Ghostscript
is this a known problem or is it just me. More
importantly does anyone know of a way to fix this, I'd like to use
8.51 instead of 6.50.
By the way, if it makes a differnece i'm currently running
[EMAIL PROTECTED] but I've encountered the same problem with all the other
asterisk builds i've tried
so how do we get this fixed, its happing to my one and only DID as well...
On 4/22/05, Me [EMAIL PROTECTED] wrote:
I had the same problem with another provider whom I got no response from
as
usual..
We had 5 or 6 numbers that worked fine and one that just quit sending
DTMF.
- Original
*
running 1.0.7 cannot identify the dtmf. IAX debug does not show dtmf being
sent to me.
Just want to know
whether any of you had this experience, and if so how that was fixed. Funny
thing is this happened on two dids and others are OK.
Cheers
DH
This is good but if your company name isn't Vonage, how do you get access?
- Original Message -
From: Norm Zimon [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 9:39 AM
Subject: [Asterisk-Users]
OK, been messing with RealTime like a week off and on, I can safely say it's
killing me!
I have dug and dug and dug to find what I am missing, no dice.
I am running the latest version of * from CVS as of about a week ago.
Call comes in from a PRI into the todd_test_1 extension, if I uncomment
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, April 20, 2005 4:42 AM
Subject: Re: [Asterisk-Users] RealTime ignoring switch=
Realtime/[EMAIL PROTECTED]
Me wrote
Yes, I downloaded via CVS then ran make then make install.
In fact I did this again last night to be sure it was installed.
Maybe I downloaded the old add on package, and it didn't come with it. I
have the latest version of Asterisk but I just pulled plain old
Asterisk-Addons from CVS.
Do I
]: res_config_mysql.c:605 mysql_reconnect: MySQL
RealTime: Everything is fine.
chase1*CLI
Yeeehaww!!
Thanks a ton, now I can move on with the show.. Please let me know if there
is anything I can do for you in return, I can't tell you how much I
appreciate your help..
I can offer you at the very least some free
Well, I bought two of these when they were first released.. They seemed like
VERY nice phones for the money except for the fact that the headset jacks
did not work at all on either device. Tried multiple headsets none of them
worked. I had to return the phones..
I also remember the buttons
I have seen folks mention FireFly softphone on the list many times. I went
to their website but could only find a version which connects directly to
their service, it did not seem configurable to use with *.
Is FireFly in fact usable with *?
Thanks!
Maybe they could start by finding the info on the lawsuit that was brought
against the last ISP that tried this. They could then forward it to their
ISP and see if that gets them anywhere.
I guess this could also get them disconnected from the only ISP available
so... Don't listen to me
I may not understand fully how any of these three features work but...
Can someone tell me what benefit there is to using RealTime instead of say
calling a MySQL database directly from the extensions.conf using the built
in MySQL functionality? Also, it looks like I could use PHP via AGI to also
Is there any better docs or step by steps other than what's in the Wiki for
Realtime setup?
We have been trying to get this running and it's driving us batty..
It seems that the switch command is totally being ignored as far as we can
tell.
We are basically just getting an error telling us
This is the asterisk output:
-- Executing Answer(SIP/202-8236, ) in new stack
-- Executing Dial(SIP/202-8236,
SIP/203|100|tTr) in new stack
-- Called 203
-- SIP/203-3c5d is ringing
-- SIP/203-3c5d answered SIP/202-8236
-- Attempting native bridge of SIP/202-8236 and
not entirely sure my PRI is 100% up even, * seems to be talking to it
because when I pull the cable it starts giving me alerts and such, the
alerts go away when I plug the cable back in.
Of course the telco is waiting for me to call them so we can test the PRI
against my equipment.. I guess they expect me
I keep hearing DTMF type beeps when on phone calls, I know this is some sort
of trait of VOIP but it's driving me nuts..
I noticed that it happens MUCH more when I am on the phone with one
particular person.
We are using SPA-2000's from Sipura on both ends.
Tonight I was looking at the CLI
asterisk-users@lists.digium.com
Sent: Thursday, April 07, 2005 12:52 AM
Subject: Re: [Asterisk-Users] Beeps during Sip to Sip phone calls
Inline...
I keep hearing DTMF type beeps when on phone calls, I know this is some
sort
of trait of VOIP but it's driving me nuts..
Not really.
I noticed
of a program/extention to asterisk that would allow
me to either text message my asterisk box or IM it from AIM on my cell
phone to allow it to call me? I've been looking with google yet can't
find anything. I don't code, so I'm SOL there, so I'm looking for
something premade. I plan on taking a class
Thanks, this looks like what I need. Setting it up looks like a career
though but hey it's free so what can you do?
:)
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial
We are in the process of ordering a Voice PRI to plug into Asterisk. Of
course we will be buying a card from Digium for this.
Question is this, there seem to be MANY options technically when ordering
this PRI (in the US) but since this is the first time ordering a voice
circuit I am clueless
We have been seeing tons of additional SPAM coming through our Modus 4
server, mostly medical stuff.
Is anyone else seeing a big increase lately? I have not seen the list for a
bit seems I was unsubscribed somehow.
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
Doh!
Wrong list, please ignore..
Sorry.. 30 lashes for me..
Todd
--
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http://www.YourOwnISP.com
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I have my main * box setup for all incoming and outgoing calls to and from
our SPA-2000's. I have now setup another * box in a different location and I
would like the SPA's to send all outgoing calls out through the new * server
but continue registering with the old * server so all incoming
Can someone point me to some docs that explain this or give me a direction
to go in. I have seen docs on this in the past but can't seem to dig em up
now when I need them.
Basically I want one Asterisk server to be the traffic cop and send some
calls directly to ATA's and some calls to another
What is your setup? Zap, ATA's etc?
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: [EMAIL PROTECTED]; [EMAIL PROTECTED]:Go Technology
Management LLC [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 24, 2005 2:36 PM
Subject:
Subject: Re: [Asterisk-Users] Damn DTMF Beeps on my calls
On Mon, 2005-01-24 at 13:38 -0600, Me wrote:
Can someone give me a clue as to why I keep hearing DTMF type beeps on my
phone calls. It sounds exactly like someone on the other end is pushing a
key on their phone but they are not!
Has anyone
Can someone give me a clue as to why I keep hearing DTMF type beeps on my
phone calls. It sounds exactly like someone on the other end is pushing a
key on their phone but they are not!
Has anyone ever heard of this before? It use to happen once in a while,
today it's been happening a LOT
We have been having WAY too many issues lately with our VOIP calls. I
suspect it may be the particular T1 we are pushing these calls out through
from our office.
Is there a decent tool out there that I can stick on the network that will
measure things like Jitter, ping times and overall
After getting zaptel from the CVS server, compiling and installing it I
type:
modprobe zaptel
and all is well. Then I type:
modprobe wctdm
and I get this:
modprobe: Can't locate module wctdm
Any idea why?
I did this yesterday but with the CVS head of Asterisk and I got by this
part without
-0600, Me wrote:
After getting zaptel from the CVS server, compiling and installing it I
type:
modprobe zaptel
and all is well. Then I type:
modprobe wctdm
and I get this:
modprobe: Can't locate module wctdm
Any idea why?
I did this yesterday but with the CVS head of Asterisk
, January 05, 2005 7:43 PM
Subject: Re: [Asterisk-Users] modprobe: Can't locate module wctdm
On Wed, 5 Jan 2005 19:09:00 -0600, Me wrote:
After getting zaptel from the CVS server, compiling and installing it I
type:
modprobe zaptel
and all is well. Then I type:
modprobe wctdm
and I get
After installing the stable version of * and the Zaptel drivers with a
TDM400 card using 1 FXO module on port 4, I start Asterisk and get this
rolling up my screen thousands of times:
Ouch... Error while writing audio data
Ouch... Error while writing audio data
Ouch... Error while writing audio
Riddell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 05, 2005 9:11 PM
Subject: Re: [Asterisk-Users] Ouch... Error while writing audio data
Me wrote:
After installing the stable version of * and the Zaptel
. Everything I've read says you
compile and install zaptel first...then asterisk. On Monday I rebooted
my server again, the just did a CVS update of zaptel. That was all the
was required.
Michael
On Wed, 5 Jan 2005 20:10:27 -0600, Me wrote:
Also, I wonder if there is some sort of issue with the fact
]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, January 06, 2005 12:26 AM
Subject: Re: [Asterisk-Users] modprobe: Can't locate module wctdm
Ronald Wiplinger wrote:
Me wrote:
10- Ran cat /proc/interrupts to make sure my card was not sharing
We will need more info on your setup.
When people call into your Asterisk system what device will they be calling
in on?
Will they call a number provided by a termination/origination provider which
is then fed into your Asterisk server using IAX or SIP?
Will they call a TDM card attached to
didn't imlepent
it yet
exten = s,46,SetVar(MACRO_RESULT=CONTINUE)
exten = s,47,Goto(50)
exten = s,50,System(/bin/rm ${ARG1}.gsm)
exten = h,1,System(/bin/rm ${ARG1}.gsm)
On Wed, 29 Dec 2004 00:35:34 -0600, Me [EMAIL PROTECTED] wrote:
Nevermind, it looks like Asterisk cmd Read is my lucky
Why not use ATA adapters? This way you can use just about any phone you
want.
Start Your Own Internet Service!
http://www.YourOwnISP.com
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 10:28 AM
Subject: RE:
I have one analog line hooked in my Asterisk box
using an x100p (I think that's the model number).
When I do this in my extensions.conf:
exten =
1200,1,playback(pls-wait-connect-call)exten =
1200,2,Dial(Zap/1/551212,20,rTt)exten = 1200,3,VoiceMail([EMAIL PROTECTED])exten =
someone tell me why
Asterisk might need fast drives vs. say 7200 IDE drives?
Next and last question is, how many simultaneous
calls do you folks figure I can run on this in the following two
scenarios:
1- All clients would be using SIP devices like
SPA-2000's and all calls would originate
Sorry about the HTML emails, on my laptop and forgot to change the sending
format from the default.
- Original Message -
From: Me
To: asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 2:01 PM
Subject: [Asterisk-Users] Sending call to analog then to Vmail after
timeout
Dorn,
Can you give me some details on this linux md driver you mentioned?
Also, you say not to scrap the SATA drives, is this because you think I can
use them with FC1 or because you think I should try Debian? I really don't
want to venture away from Fedora at the moment for a few reasons
://www.YourOwnISP.com
- Original Message -
From: Sean Cook [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, December 27, 2004 7:31 PM
Subject: Re: [Asterisk-Users] Hardware opinions?
On Tue, 2004-12-28 at 16:12 -0600, Me wrote
On Tue, 28 Dec 2004 14:20:02 -0600, Me [EMAIL PROTECTED] wrote:
Sorry about the HTML emails, on my laptop and forgot to change the
sending
format from the default.
- Original Message -
From: Me
To: asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 2:01 PM
Hetzel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 7:00 PM
Subject: Re: [Asterisk-Users] Hardware opinions?
On Tue, Dec 28, 2004 at 04:12:26PM -0600, Me wrote:
Dorn,
Can you give me some details
aftertimeout?
To: Me [EMAIL PROTECTED]
try the M option which will do a macro and will not connect the caller
unless s/he presses some button. and if no button is pressed then it
goes to VM. now remember to replay the message (to press the button) a
few times b4 going to VM otherwise they will never
Nevermind, it looks like Asterisk cmd Read is my lucky command :)
Thanks!
Start Your Own Internet Service!
http://www.YourOwnISP.com
- Original Message -
From: Me [EMAIL PROTECTED]
To: C F [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users
: [Asterisk-Users] Record() problem
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
Refer to my example on that bug note.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Me
Sent: Friday, December 24, 2004 11:06 AM
the
channel it will record is implicit and cannot be explicitly stated as one
of
the parameters.
If you want to originate and record a call automatically, you will have to
do this via AGI.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Me
Sent
Anyone know what this error message means?
**
Dec 23 23:12:31 WARNING[3031057]: chan_sip.c:1874 sip_write: Asked to
transmit frame type 2, while native formats is 4 (read/write = 4/4)
**
I see this in my CLI when I call into Asterisk and press * which should hang
up the call
Can you give me an example of how a call would end up in the timeout ext?
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Seth Remington [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday
Any idea why this:
Record(IAX2/[EMAIL PROTECTED]/5, /tmp/whatever.gsm|6|25)
Would result in this:
WARNING[3293201]: app_record.c:117 record_exec: No extension found
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
___
Asterisk-Users mailing
to 106 again. Same
thing call dies after 3 minutes or so.
I looked at the CDR entries for these two calls and two fields/columns of
the entries got me pretty curious. For BOTH calls under the billsec and
callduration fields the value was exactly 180 seconds.
This leads me to believe that somewhere
Nope, I searched the extensions.conf, sip.conf and iax.conf for 180 and
found nothing.
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
What does t mean in a CDR entry? This is in place of where the number that
was dialed normally goes. For one IAX termination provider it always has a t
instead of the number dialed. Also, we always see the word hunguup in the
same record entry. This is the provider we have set to our secondary
I have lots of these working and at least two behind NATs..
Start by setting your SPA-2000's IP address as the DMZ address on your
router. If everything works all of a sudden then that's a good start. I did
this and it least it told me that all was well with the adapter itself.
What type of NAT
It seems that all my CDR is dumping into the Master.csv file. There is a way
to create per user/extension CDR but I have looked endlessly in the Wiki,
docs, README.CDR, mailing list archives etc.. I can't seem to find a way to
do this..
Any help would be appreciated.
Thanks!
--
Start Your Own
I would like to setup call confirmation so that the called party has to
press a key to accept the call. There seems to be an Asterisk feature to do
this with Zap channels where you place a c in the dial string. I want to
do the same thing without re-inventing the wheel with IAX and SIP
etc works great with
asterisk. I recently purchased from www.qualvoip.com (they also provided me
sample configuration files for asterisk).
Kevin
- Original Message -
From:
Shawn
Dillon
To: [EMAIL PROTECTED]
Sent: Wednesday, December 15, 2004 8
I have been most impressed with iax.cc lately.. Only been with them a few
days but so far, so good!
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Mike Diehl (Encrypted email preferred) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
which will allow me to prioritize voice traffic on this
link?
I can't change the T1 router to something that supports QOS because it has
certain redundant features with an ISDN line which are needed.
No commercial interest, just a satisfied customer. . . .
NetEqualizer from APConnections http
Here is the situation:
A T1 router going into an office which then plugs into the firewall box then
into the switch.
None of these devices support QOS..
Is there some sort of box/device that I can place between the T1 router and
the firewall box which will allow me to prioritize voice traffic
: Monday, December 13, 2004 1:56 AM
Subject: Re: [Asterisk-Users] Follow Me Music on hold
Me wrote:
OK, I have an extension setup with a follow me like so:
;Operator Going to Sue first, then Mary
exten = 0,1,playback(pls-wait-connect-call)
exten = 0,2,Dial(SIP/103,20,mTt)
exten = 0,3,Dial(SIP
Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad.
Thanks!
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It seems that this is now fixed!
Looks like it was the NAT Keep Alive setting which needed to be set to
yes in my case.
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Me [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 12, 2004 10:45 PM
Subject
OK, I have an extension setup with a follow me like so:
;Operator Going to Sue first, then Mary
exten = 0,1,playback(pls-wait-connect-call)
exten = 0,2,Dial(SIP/103,20,mTt)
exten = 0,3,Dial(SIP/102,20,mTt)
exten = 0,4,VoiceMail([EMAIL PROTECTED])
exten = 0,5,Goto,t|1
This works well except
I had a Grandstream 286 at my home hitting my Asterisk box at the office,
all worked well and I received phone calls fine until the device just up and
died.
I replaced this unit with an SPA-2000 because I have been impressed with the
Sipura devices and decided to use them for most of my needs in
Personally I find the ATA adapters to be the most versatile, your mileage
may vary though. When you need more extensions you just buy more ATA's, no
need to tear up the * box or take it down etc.
Buying IP phones is OK but you are limited to IP Phones only. With the ATA's
you can buy ANY phone
I had a Grandstream 286 at my home hitting my Asterisk box at the office,
all worked well and I received phone calls fine until the device just up and
died.
I replaced this unit with an SPA-2000 because I have been impressed with the
Sipura devices and decided to use them for most of my needs
with Termination Providers?
Indeed they do - but if you want numbers, you need to say where you are -
there is no point our company supplying you with UK numbers or toll free,
if you actually US people to call them!
- Original Message -
From: Me [EMAIL PROTECTED]
To: Asterisk Users Mailing
I saw them too and they looked pretty good. I assume you can buy the minutes
and use them for whatever you want.
Only issue I have with them at the moment is that their ping times don't
seem great from where I will be setting up our initial server.
I may setup an account with them for testing
Dead for me too.. I am in the US..
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: David Uzzell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, December 02, 2004 12:41 AM
Subject: Re: SV
] Experiences with Termination Providers?
Me wrote:
I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the following:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
Mostly interested in US to US for now but interested in all areas, I was not
aware I was restricted to looking for a provider in only certain areas. Most
of the termination providers I have dealt with so far offer calling
worldwide.
Thanks,
Todd
--
Start Your Own ISP!
http://www.YourOwnISP.com
I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the following:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
-Reliable/Redundant network and
On this page in the Wiki:
http://www.voip-info.org/wiki-Asterisk+ZAP+Channels
This text exist:
*
If the letter c follows, then Answer Confirmation is requested, in which
the call is not considered answered until the called user presses #.
*
Question:
From what I
The thing is, why run it on Windows.. Even though there is a Windows version
now it's not really a Windows version is a Linux version running on a
version of Linux that will run on Windows.. YUCK.. That's like taking a
Cadillac engine and putting in a Yugo just because you feel more comfortable
Give us your extensions.conf and we may be able to help you
___
Not sure if you wanted all of it but here it is with my ID's and domains
changed of course.
*
[general]
static=yes
Any ideas on this folks? I am kinda stuck without it..
Thanks for any help you can provide..
Todd
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Me [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday
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