By UK standards that's a pretty good salary.
Bear in mind that there is no real 1:1 parity in IT salaries. In the US
we earn significantly more for our IT efforts than in the UK.
To give you an example, when I moved from London to New York I got a 4
fold pay rise in real terms for doing
I would second that.
If you don't set a dial string in your handset then it waits for N
seconds before submitting the call. Pressing # will force an immediate dial.
Mark
On 11/04/2010 07:19 PM, Cary Fitch wrote:
Watch the console as you dial. Dial the number and “#”. The ring
should be
They say confession is good for the soul. Perhaps they are offering a
phone in confessional service?
Unfortunately the business of the church often flies in the face of
the business of the Church.
On 03/29/2010 07:48 PM, Alex Balashov wrote:
Sounds like the church has strayed from its core
Probably worth discussing this over on the AstLinux list as they are all
about embedded Asterisk running on machines like this.
On 09/22/2009 09:48 AM, Danny Nicholas wrote:
I was going to dismiss this, but it does offer an interesting possibility;
Since it can boot Debian ARM from an SD card,
/asterisk-users
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Mark Phillips, G7LTT/NI2O
Randolph, NJ
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to fill an
hour of her time. So far we have about 25 minutes.
Le me know off list.
Thanks
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Mark Phillips, G7LTT/NI2O
Randolph, NJ
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that comes in when I'm on my cell get sent to VM or does it ring the
follow-me group again?
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Randolph, NJ
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AstriCon 2008 - September 22 - 25 Phoenix
Damn!!! Beat me to it ;-}
As an Englishman now living in New Jersey (strangely nowhere near an
exit) I have to say that the local idiom and accent leaves a significant
amount to be desired.
Terms like New Joisey, Shuwa ,wadder, badderies,
congradulations etc make me wonder if I'm in an English
Hi all,
I have a cheapskate customer whom wants to leverage some cheap
all-you-can-eat VoIP connections rather than pay for a per minute
provider.
On the inbound side I think I have a solution in that I can activate the
call forward on busy option with his provider (some noname white label
Sounds to me like inband vs rfc2833 issues.
I found that one has to use the same codec throughout in order to make
DTMF function and then use inband. This in turn forces you down the road
of alaw or ulaw codecs.
On Tue, 2007-06-26 at 18:01 -0500, JR Richardson wrote:
Hi All,
I have
Great! Another one. With such a catchy name too!
On Tue, 2007-06-26 at 01:42 +0200, lenz wrote:
Hello list,
AstPligg is a new Digg-like website devoted to * and VoIP news.
At the moment, it's in beta stage and very basic - no fancy custom
templates. It allows posting new stories, comments
Hi folks,
I'm experimenting with Heartbeat and whilst I have it running in an
active/standby configuration I cannot get Asterisk to perform properly.
I'm able to start the asterisk software (I imported the aterisk start
file from /etc/init.d into /etc/ha.d/resource.d) with the heartbeat
software
Hi all,
Has anyone tried using Asterisk as an SCCP client?
My company has just signed up a 2 year agreement with M5 (fools!!) but
are having intellectual issues with things like intra office phone calls
and voice mail etc. They suddenly realized after M5 was installed that
ALL their calls go out
Nokia N95 available via ATT/Cingular for $795 with a 2 year contract. It
was advertised in the New Jersey Star Ledger this morning.
Mark
On Thu, 2007-05-24 at 18:42 +0500, Rizwan Hisham wrote:
Hi all,
sorry to ask you something not related to asterisk, but i really want
to know whether the
Hi all,
Has anyone tried using an ITSP that utilizes SCCP as it's prime mover?
Would it actually be possible?
A customer of mine had M5 installed yesterday and they are already
disliking the idea that their provider is in possession of all their
VM's and that they have to go out to the Internet
Without seeing your config files my guess would be that this is
something to do with a bad codec negotiation.
I'd bet that your IAX phone is using ulaw and your DID provider is using
something else like G729.
Mark
On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote:
HI
I've configred an
Hi all,
I have a surplus Sangoma 10 port FXO card for sale. This model could be
upgraded to 12 ports or even changed to FXS or a combo of FXO/FXS by
changing the grand-daughter cards (each card supports 2 lines). You
could also downgrade the card by removing any or all of the daughter
cards.
I'm
Why not. Users by stuff too!
On Fri, 2006-11-24 at 11:40 -0800, Anthony Rodgers wrote:
Please don't cross post FS items to *-users - that's what *-biz is for.
CP
On Nov 24, 2006, at 10:45 AM, Mark Phillips wrote:
Hi all,
I have a surplus Sangoma 10 port FXO card for sale
Hi folks,
I have a Sangoma A200 10 port FXO card for sale.
US$500 secures plus shipping.
Thanks
Mark
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I don't think that that there's any way around this. At some point you
require human intervention.
Perhaps the only way to do it would be to set up some sort of timer.
After x seconds if you don't get a key press Asterisk moves the call to
it's own VM?
On Wed, 2006-10-04 at 07:00 -0200, Daniel
You don't need to restart Asterisk. Just do a reload app_voicemail.so
On Wed, 2006-10-04 at 06:45 -0500, Jordan Novak wrote:
How is the best way to add,clear mailboxes and change passwords for
voicemail. I am guessing you need to remove the conf entries for the
mailbox restart asterisk and
What do yo mean by fails?
If you don't if one party doesn't have the preferred CODEC Asterisk will
fall back to the next preferred CODEC and so on until a match is found.
Can't help you on the licensing thing though. I guess no one wants to
touch it since Digium's stance seems to be that you
What tools are you using for this?
I'm sure you are aware of SIPp but wondered if you had anything else?
Mark
On Thu, 2006-09-07 at 21:41 +0200, Matt Riddell (IT) wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
RR wrote:
Hi matt,
sorry this might be a stupid question but is a
Sounds to me like you don't have a proper connection with Stanaphone.
The only time you'll get these problems is when they cannot contact you
to forward the call to your system.
Double check you firewall settings. They need to be able to reach your
system on port 5060UDP (assuming SIP) as well as
SPA941's and 7960's
On Thu, 2006-07-06 at 12:44 -0700, Shaun wrote:
What brand/model phones are you using.
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Perhaps the BT crew are all on a drunken rampage along Sochiehall
Street?
On Mon, 2006-07-03 at 15:14 +0100, Colin MacMillan wrote:
Hello,
For some reason I can't call Scotland from London ...
The details:
Asterisk v. 1.2.9.1
ISDN2 Interface - Junghanns card with BRIstuff 0.3.0-PRE-1q
T-Mobile do GSM, GPRS and EDGE and not GSM only as stated below.
Devices connected to their network typicly use GSM but may use GPRS if a
data plan is subscribed. Edge is available o those that have both an
Edge device and a data plan.
Not that I'm a T-Mo reseller or anything ;-}
On Mon,
Actually this is an Elastic Impact. Throwing an object at another object
as suggested below could cause the kinetic energy possessed by the ball
to be diverted thus causing the ball to travel in a different direction
after impact.
This type of impact is commonly seen when insufficient kinetic
Hi folks,
Verizon hand just installed FIOS to the side of my house. Anyone know
anything about their VoiceWing offering? Is is a SIP offering? Their
technical staff can only tell me that it's a VoIP service.
Mark
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Yet another set?
I get about 50 downloads a week for mine.
Mark
On Tue, 2006-06-06 at 22:27 +0100, Steve Kennedy wrote:
I'd like to announce that the UK Male English Voices are now up on
http://www.tel.net/
There's a complete set of base sounds and additional sounds (it should
be complete
Hi Gabriel,
This phone does not have a SIP image available for it. It does use a
modified version of H323 but you should be able to use it with Asterisk
if you have something like Open323 installed.
You'll need to install a TFTP server onto your network which the phone
looks for to find it's
Ig nore my last post. I had not seen this posting
On Tue, 2006-06-06 at 22:33 +0200, Henk wrote:
Have a look at the attached link.
http://support.avaya.com/japple/css/japple?temp.documentID=283920temp.productID=107755temp.bucketID=108025PAGE=Document
Henk
Hi folks,
Anyone got a gently used working T1 card I can have?
Can pay by CC, check, cheque or Paypal.
Thanks
Mark
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Not quite.
It refers to the contiguous 48 States that make up the US mainland.
Alaska and Hawaii, whilst States, are separated by either another
country or large amounts of ocean.
Places like Key West whilst technically are over 50 miles from the
mainland are considered part of the Lower 48/US48
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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Hi Paul,
Asterisk often uses a proxy for its calls. What kind of proxy do you
have?
Also, If you have the server setup for nat=yes in the [general] area
then ALL calls will get nat'd regardless of their locality. The best
place to put this stement is in the relevant part of the sip.conf file
As I understand it, the device uses either GSM or VoIP to access the
carrier? Which cell phone carrier supports GSM and VoIP in the EU?
They've been punting this thing around the shows in the US for a couple
of years now but none of the carriers support it. With GSM having such
blanket coverage I
Hi Peter,
I don't see any codec allow=blah statements. If your end user has
something like
[gradwell]
disallow=all
allow=gsm
Then you'll be forced to send them a GSM coded call.
Why not force the codec at your end by only supporting one? If the
customer then transcodes the call when it gets
Don't ya just love living in this technological backwater they call the
USA? DECT technology was released almost 20 years ago. In most of the
world it's been and gone.
Anyone in the UK or Hong Kong remember Rabit and having to find a Green
Dot Hot Spot by the train station or post office? When
I wonder what they think VoIP is? Are they just port blocking? Could
they be doing packet inspection? Do they think all UDP trafic is VoIP?
On Mon, 2006-05-22 at 11:14 -0400, Julio Arruda wrote:
From what I understand, T-Mobile UK just announced they would block
VOIP earlier this month,
I know what he's done.
He's installed my Alison Keenan wav files without converting them. Try
downloading the sln files instead.
BTW, G723 and G729 files going up tomorrow.
Mark
On Tue, 2006-05-23 at 00:49 +0200, Patrick wrote:
On Mon, 2006-05-22 at 23:11 +0100, Robbie Hughes wrote:
[snip]
Obviously a Radio 1 listener.
2) I was surprised to find that I didn't like the results.
This is a purely personal thing, but I found
Alison Keenan's delivery too redolent of a England that is
gone. I instantly felt like a child again, being told slowly and
clearly what to do.
and don't forget to practice safe IAX ;-}
Mark
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Mark Phillips [EMAIL PROTECTED]
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--
Mark Phillips [EMAIL PROTECTED]
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Hi folks,
It seems that BV has messed it up yet again.
I noted this weekend that any call going in or out had no incoming
audio. All my other SIP providers seem to be OK. Is anyone else having
this problem?
Perhaps it's time to move on. What providers do you recommend that
provide unlimited
Hi folks,
I have British comedienne, Alison Keenan (another Alison!) coming in on
Saturday afternoon to record the Asterisk prompts for me. Alison speaks
with a posh boarding school accent. Finally we'll have a free British
English female voice bank.
As I have her in my studio (yeah right; it's
I think it is correct. Isn't that why they call it a Smart Jack? I've
only ever seen a regular cat5 cable used from the Smart Jack to the
device (router/PBX/CSU/DSU/whatever).
I believe the point of the smart jack is, amongst other things, to allow
for the use of readily available cables.
I
Likewise here.
Using a 10 port FXO card and no problems detecting remote hangup. I'll
grant you it can be a little slow sometimes however.
On Mon, 2006-04-24 at 16:54 -0500, Rich Adamson wrote:
Mike Garey wrote:
As far as I can tell, after discussing this matter with other asterisk
users
Just for shits and giggles, have you tried using a cross over cable? I'm
not saying it's gonna work because everything I read says you're doing
the right thing but it's worth a try.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Dmitry Ivanov wrote:
Hallo!
Anyone tried connect PC
Erm ... isn't this what a conference does?
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Leonardo (listas) wrote:
I will implement a SIP application and I'm considering using Asterisk
for mixing the media streams (audio). Does anybody know if Asterisk
supports or contains a RTP mixer?
I think I've asked this before and think that Matt had said something
about this.
Is there an LCDproc client for Asterisk available and if so how can I
get a copy please.
Thanks
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
___
Do you have the right cable?
You need a cross-over T1 cable and NOT a cross-over ethernet cable that
people commonly try. This should satify the electrical requirements and
turn the lights green.
You're on your own with the rest.
I do have a question however; why are you now speaking SIP to
Actually no.
As far as I understand it, the receiving station gets to dictate the
codec used. You call and offer up your list. He selects his preffered
from your list and off you go.
in your case you will always have gsm from 12 becasue 2 has a
prefference for GSM.
Try it back the other
: Restricting
registration for peer 'ext1' to 60 seconds (requested 300)
-Original Message-
From: Mark Phillips [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 16, 2006 17:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How do I install speex
Did you rebuild asterisk after your speex install?
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Jesus E Zepeda wrote:
Hi, everybody:
I enabled speex in my asterisk box (iax.conf), but no phone call went
throug. At the asterisk console, I used the show modules command and
it did not
If you did a make install with speex then everythings where it should be.
Just do a make; make clean with asterisk and all will be fine.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Jesus E Zepeda wrote:
Huuu! I never expected you had to recompile asterisk to add a codec. But
if that
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: Thursday, February 09, 2006 5:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk native sounds now available!
Yes you can copy them into the same
to try
out. Not a bad price at $79.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Alex Barnes wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: 07 February 2006 19:23
To: Asterisk Users Mailing List - Non
Is a panoplie legal in Wales? I thought they did away with those at the
same time as the Wooly Mountainside Brothels?
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Wilson Pickett wrote:
I've been looking for someone whom speaks both with a Welsh accent and
also the language.
Check
Yes you can copy them into the same directory as the current files. Kris
recommends that you move your existing files for safety only.
The mode (ULAW, GSM etc) is selected by Asterisk depending upon what
mode the current caller is using.
Have you noticed that you don't have to put a file
Kirs et al,
I did this already. It's on my website. Your most welcome to use them
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Kristian Kielhofner wrote:
Alex Barnes wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
I dunno about your provider but I know that 2 of my 3 MCI PRI circuits
have no 911 abilities. MCI tells me this is becasue I have no local
dialing plan on them.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Michael Collins wrote:
911 **should** work on a PRI. If you are getting a
One problem I can see is that you're not using the keys that come with
asterisk.
Mine (which works!) looks like this
iax.conf
register = user:[EMAIL PROTECTED]
[iaxfwd]
type=peer
context=from-fwd
permit=65.39.205.0/24
auth=rsa
host=iax2.fwdnet.net
inkeys=freeworlddialup
disallow=all
Erm ... sorry. That should read Kris et al
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Mark Phillips wrote:
Kirs et al,
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The same 7 sound file is used to indicate both time and quantity. The
sound file could be easily recorded to say sept heure but then every
time the VM system tells a user that they have 7 messages they'll hear
something like vous avez sept heure notification (excuse my schoolboy
French).
I've come across this in my dealings with my customers in Toronto. As an
Englishman I find it most infuriating. French is after all, the most
hated language in the world from an Englishmans perspective ;-}
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Derek Whitten wrote:
Colin
Aha!! why didn't I think of that.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Gonzalo Servat wrote:
On 2/6/06, Mark Phillips [EMAIL PROTECTED] wrote:
A customer of mine wants an IVR where the first 3 choices are
1 English
2 Spanish
3 French
I can build the IVR but how do I get
Try adding insecure=very to the guest user account in iax.conf. This
should not do a user/pass challenge on the incoming call.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
kevin ling wrote:
Not sure answer your question? Try to write some html code and let user
register the username
I forgot to add that you must have an IAX acount with FWD. A regular SIP
account won't let you then use IAX. You have to register for it.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Mark Phillips wrote:
One problem I can see is that you're not using the keys that come with
asterisk
.
-Original Message-
From: Mark Phillips [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 07, 2006 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] change languages from an IVR
I've come across this in my dealings with my customers in Toronto
A customer of mine wants an IVR where the first 3 choices are
1 English
2 Spanish
3 French
I can build the IVR but how do I get the system prompts to then speak
the selected langauge. For example, a caller has selected Spanish and so
is routed to the Spanish part of the IVR. At some point he
Whilst it can be downloaded I find that a paper copy is easier to read.
I bought it for that reason alone. I also find it's a usefull addition
to my tool box. I can't always access the net whilst on site. If I get
stuck doing something I can look it up in the book.
Mark, G7LTT/KC2ENI
It does indeed.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
James Ronald wrote:
Does the printed version have an index?
-- JR
Whilst it can be downloaded I find that a paper copy is easier to
read. I bought it for that reason alone. I also find it's a usefull
addition to my tool
This looks like the solution.
I'll let you know how I get on.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
[EMAIL PROTECTED] wrote:
Hello,
MP Can I couple this to the sound card in the Asterisk server and then have
MP it play into the MOH? If so how?
Yes, it's possible. I've tried
=Asterisk+config+musiconhold.conf)
and sox with the alsa pseudo-filetype, and output to stdout with the
correct bitrate and samples... see the sox manpage for instructions.
Untested, but I think that should do the job for you...
Mark Phillips wrote:
I thought this had been around before but I can't
I thought this had been around before but I can't seem to find anything
about it.
I have a customer whom prior to upgrading to Asterisk invested in one of
those boxes that plays your company sales campaign into the MOH port on
your key system.
For reasons of message maintenance he wants to
Throw it in the trash now. There's next to no support for these. No
firmware upgrades. The are VERY SLOOW in responding to network
calls too.
All in all not a very astute purchase. I should know; I've had 5 of them.
I use the UTStarcom F1000 currently. Much better but still not good.
An example of this would be Outcall Voice Mail?
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Danish Samad wrote:
Hi,
In a normal PBX environment a user usually calls in and IVR's are
played according to a predefined dialplan.
Iam trying to develop an application where asterisk
Most often the simple addition of nat=yes in the relevant sip.conf
stanza is all that's required to make a remote SIP phone work from
behind a firewall.
for example
[2201]
user=blah
secret=blah
auth=blah
allow=blah
host=dynamic
nat=yes
I've been running 4 remote SIP phones across the
Contact me off list if interested.
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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This is a great idea!
You could have an IVR presented by a computer generated figure. You
could play viewzak to folks on hold. Or how about the company promo
reel when waiting for you turn in the call center queue?
I'm loving this idea!!
In a previous life I used to be a video editor for
It has to be said that Eid is a funny and possibly suspect celebration
though.
As I understand it (from one of my Muslim underlings) 3 Mad Mulahs have
to look for a particular phase of the moon. When they see this phase
they declare the start of Eid. They apparently get 3 nights in which to
Are they configured for inbound calls? If so how?
Usually the telco sends the last 4 digits of the called phone number
down the line. This means you'll need an exten=blah setup in the context
that handles the T1.
Hope that helps.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
David
They're not? They have no business in an open source world then ;-}
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Douglas Garstang wrote:
Not everyone is a C programmer extraordinairre.
-Original Message-
*From:* Alyed Tzompa [mailto:[EMAIL PROTECTED]
*Sent:*
Hi all,
Anyone got any VoIP traffic shaping rules for m0n0wall that they could
let me look at please?
Thanks
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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Asterisk-Users
+++
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
+++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
fax - 716.630.1548
Mark Phillips wrote:
Judicous application of my Staples Easy Button reveals this to be a
no name special I Googled it and found the device badged under
Take a look at voipsupply.com. They have a number of devices that allow
a wireless phone to be connected to a * server. One of their units has a
built in ATA and another is compatible with X-Pro/X-Lite.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Dan Elder wrote:
I've been
Judicous application of my Staples Easy Button reveals this to be a no
name special I Googled it and found the device badged under Ipeya,
BossLAN and a whole host of others.
Until recently it was on Voipsupply.com too.
This is one of the devices that was recently discussed a being a sucky
Yes they do use Asterisk for some of their facilities.
However, Alison is a contractor and so whomever pays her money gets her
voice.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Joe Pukepail wrote:
I heard on the radio about 1-800-FREE411 and tried it out, I was very
suprised to
' for another voip project about 6 months ago.
I don't remember/care about the details but that was the story.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: Thursday, December 29, 2005 6:05 PM
This would not be the chosen method.
Also, you have connected an fxo device to an fxs device. This will
produce the results you have encountered. Connecting 2 fxo's or 2 fxs's
together would not produce anything as you have discovered.
The prefered method would be via a cross-over type T1/E1
sound is all broken? WTF is that meant to mean. Does it play or
doesn't it?
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Zeeshan wrote:
Hi,
When I call to my asterisk server, voice prompts play ok but when it
goes to music on hold, sound is all broken. Why is that, is there
Assuming its a SIP based device
[110001]
user=something
allow=whatever
callerid= lateef
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Code Lover wrote:
Hi all,
How i can change the CallerId format in plan id?
for the example i can see the value of CALLERID variable like
lateef
This is a stable, well used firmware version. It fixes a load of faults
that have plagued users. You should be fine
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Tomislav Parcina wrote:
I have Grandstream Budge Tone 102 with Software Version:Program--
1.0.5.18Bootloader--
4 T's for 100 users? That's almost a line each. Talk about overkill and
expense!
Whatever happened to the 3:1 rule of thumb? That would require 33 lines.
Obviously this would be a little difficult to produce so 2 T's for 48
lines would be what I'd install.
If you have 100 users and they are
This is the how long is a piece of string question.
It all depends on the hardware Asterisk sits on, the codecs in use, the
dialtone provider (SIP vs IAX vs T1/E1) etc.
Do a wiki search and you'll find some examples of what folks have found.
As for originate on one and terminat on another;
I don't know if the registration bug has been fixed but I've not seen
the registration problem for some time.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Bob Goddard wrote:
On Friday 23 Dec 2005 08:03, Tomislav Parcina wrote:
I have Grandstream Budge Tone 102 with Software
I'm not sure what you mean here but you do realise that BV don't allow
you to set the caller ID. They fix it in there system.
I guess it gets displayed s whatever the minimum wage keyboard operator
enters it in as.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Paul wrote:
The logs
What's wrong with us that celebrate Kwanza?
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Dmitry Ivanov wrote:
On Friday 23 December 2005 10:22, Mauro Zanin wrote:
Hi everybody,
no issues this time. Only stopped to say: Merry Christmas and Happy
New Year.
Yes, Merry Christmas,
Hi Folks,
I've just built myself a m0n0Wall based around a WRAP board and whilst
it work really well for everything else I'm having some issues with
Asterisk's NAT abilities.
Here's my setup,
A bunch of hardphones (various types) littered around the house.
SPA-3000 handles the house POTS
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