How exactly does Asterisk provide E911 service??
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Hi,
What do I need to do to get Asterisk to allow me to use codec G-726?
I've already tried allow=all in my sip.conf config.. didn't work...
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Oh this is sad.. I'm familiar with radius.. and was hoping to be able
to use asterisk with freeradius to be able to do call accounting and
billing.. so you're telling me this is now not a good idea?
Am I better off (for now) parsing the csv report each month?
On Thu, 17 Mar 2005 11:00:09 -0600,
Hi,
I have one phone on my network that just keeps ringing (when I call
it) and does not go to voicemail.
If the person there is on the phone, and someone calls it they get the
busy message, but they never seem to get the 'unavailable' message...
instead it will just ring and ring and ring... any
Not really anything of use.. but here they are:
voicemail.conf:
201 = 5929,Matt Hoppes,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes
202 = 202202,Michael Eck,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes
203 = 203203,Matthew
Kiessling,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes
sip.conf
extensions.conf:
; Asterisk Management Portal (AMP)
; Copyright (C) 2004 Coalescent Systems Inc
; dialparties.agi (http://www.sprackett.com/asterisk/)
; Asterisk::AGI (http://asterisk.gnuinter.net/)
; gsm (http://www.ibiblio.org/pub/Linux/utils/compress/!INDEX.short.html)
; loligo sounds
Hi,
Does asterisk have in itself an STUN server built in? Or do I need to
set one up seperately? And if that is the case, what is recommended
for use with asterisk (to allow VOIP users behind nats to connect to
my VOIP servers)
Matt
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If I only want to give my sip users say local calling where do I put
that in the sip config?
I have the contexts setup.
[outbound-local]
exten = _NXX,1,Macro(dialout-default,${EXTEN})
exten = _NXXNXX,1,Macro(dialout-default,${EXTEN})
[outbound-tollfree]
exten =
Got it.. thanks that worked...
On Fri, 18 Mar 2005 09:12:34 -0600, Scott Nelson [EMAIL PROTECTED] wrote:
Matt wrote:
If I only want to give my sip users say local calling where do I put
that in the sip config?
...
and the sip.conf looks like:
[200]
...
context=from-internal
Has anyone used asterisk as a simple voip server? (I'm sure its been/ing done).
If so... how did you provide 911 service? Did you setup different
contexts and put sip phones in those contexts per county?
Also, is it possible to put a phone into multiple contexts?
For instance:
Hi,
In the current asterisk release 1.0.6 does G726 currently support
16/24/32/40, or are we still only at 32kbps?
Are there any plans to allow variable packetization rates per sip
device for applications like faxing over voip, etc?
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Hi,
If I have a PRI card in my asterisk server and have VoIP dial-tone
from Level3 over Ethernet, how do I go about setting up a routing
table to route all calls out over level3 with the exception of:
Any calls which would be local on the PRI (certain exchanges which I
will program in), and
most helpful thanks! :)
On Sat, 19 Mar 2005 16:01:31 -0500, Tyler [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
should tell u everything u need.
tf.
On Sat, 2005-03-19 at 15:51, Matt wrote:
Hi,
If I have a PRI card in my asterisk server
:06 -0500, Matt [EMAIL PROTECTED] wrote:
most helpful thanks! :)
On Sat, 19 Mar 2005 16:01:31 -0500, Tyler [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
should tell u everything u need.
tf.
On Sat, 2005-03-19 at 15:51, Matt wrote:
Hi
Which script in [EMAIL PROTECTED] are we talking about? I can't say I've
ever seen it!
On Fri, 18 Mar 2005 21:29:48 -0500, Steve Prior [EMAIL PROTECTED] wrote:
Wolfgang S. Rupprecht wrote:
[EMAIL PROTECTED] (Steve Prior) writes:
The recorded prompts by Allison are more in line with the
there.
I think like about 50 people have downloaded it since it got put up a
few weeks ago.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Sunday, March 20, 2005 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
I'm using a sipura SPA-841... Asterisk seems to be silence away (in
that it doesn't send data if it's silent)... I've set the sipura
device to be silence aware... but it still seems to send data even
when I hit mute.. anyone have any experience with this device or any
thoughts?
Ahh n/m found it:
http://sourceforge.net/forum/?group_id=123387
There definatley should be a link for that on the main [EMAIL PROTECTED] site!
On Sun, 20 Mar 2005 15:29:42 -0500, Matt [EMAIL PROTECTED] wrote:
Hrmm.. do you happen to have the URL off hand? I looked at the
[EMAIL PROTECTED
Does anyone have an example for using a live mp3 shoutcast stream with
mpg123 for hold music?
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. change your MoH class to 'live' for this example and you're done.
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 21, 2005 6:53 AM
Subject: [Asterisk-Users] mpg123 home
I can't seem to get the message on hold class to work for anything but
default.. it works if I specify default but if I specify anything else
it hangs up on me:
== Spawn extension (from-internal, 9472, 3) exited non-zero on 'SIP/200-9f2c'
-- Executing Macro(SIP/200-9f2c, hangupcall) in new
for about 6 months, maybe I just
didn't notice a problem. As far as I know there has been music playing when
people are being put on hold every time.
- Original Message -
From: Ken Godee [EMAIL PROTECTED]
To: Matt [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Really? I just tried it and WHEN it's working.. it is streaming..
and even when I hang up it keeps mpg123 up and running in the
background.
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Does anyone have any experience with asterisk and this radius module?
http://appradius.minitelecom.org/
If not, what radius module is recommened, for tracking SIP phone calls
for things like billing per phone?
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Hi,
With everyone other that who uses Asterisk.. what is the best solution
you have found for billing VoIP users? Radius? Just parsing CDR
reports nightly?
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PROTECTED] wrote:
At 10:31 AM -0500 on 3/22/05, Matt wrote:
Does anyone have any experience with asterisk and this radius module?
http://appradius.minitelecom.org/
If not, what radius module is recommened, for tracking SIP phone calls
for things like billing per phone?
Here's another Radius
Install asterisk * home.. it will quickly allow you to set things
up... as well as creating the HG.
On Tue, 22 Mar 2005 13:27:29 -0700, Gordon Anderson
[EMAIL PROTECTED] wrote:
Hello all,
Quick description of scenario:
Would like to be able to plug in an analog line to a Digium
Well this is true.. how reliable is that though? I know even with
dialup we SOMETIMES will miss a call accounting packet because they
are sent UDP
On Thu, 24 Mar 2005 00:17:31 -0500 (EST), Greg Boehnlein [EMAIL PROTECTED]
wrote:
On Tue, 22 Mar 2005, Matt wrote:
Hi,
The reason I
Indeed.. there is no $40 cancellation fee unless you fail to return
their ATA.. then they charge you and it's yours... what you think
those devices are free?
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I currently have the following outbound-local config in my setup
I can call SOME of the numbers (like 337, and 998, and
323).. but when I try to dial say like 601 I get a 404.. any
thoughts, I can't see any difference in the config.
Also, I seem to be able to dial any number
Because it seems if I dial 9 before the number all of my dial rules
get ignored... but I'd like to avoid the 9 anyway.
On Thu, 24 Mar 2005 11:08:02 -0500, Matt [EMAIL PROTECTED] wrote:
Ok.. apparently if I dial 9601 then it works.. and 9 is set as my
outside line digit... but I seem
, 24 Mar 2005 09:34:38 -0600, Henry Devito [EMAIL PROTECTED] wrote:
What do you get for an output from the CLI? Is the 9 being stripped?
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent
Hi,
This is probably a slightly odd question
Is there anyway to decrease packetization? I'm using voip over a
wireless network, and framerate is extremely important!
Is it possible to reduce the framerate? Right now it's using about 90
frames/sec in to the phone and 40 frames/sec out from
transmission of audio?
On Thu, 24 Mar 2005 20:21:59 -0500, Matt [EMAIL PROTECTED] wrote:
Hi,
This is probably a slightly odd question
Is there anyway to decrease packetization? I'm using voip over a
wireless network, and framerate is extremely important!
Is it possible to reduce the framerate
I have the following config:
[app-callforward]
; dialed call forward app - forwards calling extension
exten = _*72.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:3})
exten = _*72.,2,Answer
exten = _*72.,3,Wait(1)
exten = _*72.,4,Playback(loligo/call-fwd-unconditional)
exten = _*72.,5,Playback(loligo/for)
I've enclosed by config... I've tried everything from lowering the
tx/rx gains.. to toying with 32/64/128 echo canceling taps... at 256
echoing is really bad...
I've even tried recompiling the zaptel driver with the MARK2 super
echo canceling support...
I still have a very slight echo that I
this http://www.xgforce.com/loadbalancer.html might help too at cheaper
price.
Matt
- Original Message -
From: Andres [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, March 27, 2005 10:37 PM
Subject: Re
Web Meetme is now installed by default and the
meetme2 application is no longer needed.
What does this mean exactly? Does this use the regular meetme as
opposed to the meetme2 we had to setup before?
On Mon, 28 Mar 2005 17:35:37 -0800 (PST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
We had
you can use dual T1, each on a separate pbx. and use a load balancer for
fail over. see http://www.xgforce.com/loadbalancer.html for affordable
models.
Best Regards
Matt
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Hi,
What happened to asterisk @ home 0.7 that the dialout-default macro no
longer works?
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How would I go about giving sip users multiple contexts? For instance
right now I have them all in: from-sip-internal
Is there a way I can (for sip users) also include say my [dial-911]
[dial-local] and [dial-longdistance].. bearing in mind that I want to
have different sips allowed to do
...
Then in that context, include the features you'd like for each group,
and give each sip user the correct context.
Julian J. M.
On Wed, 30 Mar 2005 09:30:16 -0500, Matt [EMAIL PROTECTED] wrote:
How would I go about giving sip users multiple contexts? For instance
right now I have them all
:21:50 -0500, Matt [EMAIL PROTECTED] wrote:
Right,
I understand the logic behind this, and normally this is what I'd do..
but in this particular instance.. some users are going to have configs
that are different then what others have... I guess the answer is NO..
you can not have multiple
) then create a context for each group, and include into each
of those contexts what you want to let them do.
hope this helps.
bye,
M.
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent
buy 2 load balancer to failover between themselves.
Best Regards
Matt
- Original Message -
From: Mitchel Constantin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 29, 2005 11:34 PM
Subject: Re: [Asterisk
Hi,
how can I get all the phones to enable call waiting by default instead
of having to dial *70 on each one to activate call waiting?
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Actually I've noticed advanced options don't work in 1.0.6 either!
On Thu, 31 Mar 2005 16:52:03 + (UTC), Tony Mountifield
[EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote:
Folks!
I want to let everyone know that I have been
interesting thought, once I got time, I might setup a forum for folks of
asterisk.
Best Regards
matt
- Original Message -
From: Matt Ryanczak [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 31, 2005 10
?
On Thu, 31 Mar 2005 11:03:12 -0500, Matt [EMAIL PROTECTED] wrote:
Hi,
how can I get all the phones to enable call waiting by default instead
of having to dial *70 on each one to activate call waiting?
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Asterisk
I apologize for not providing enough info My
question though is not how to get my provider to provide it... how do I
enable call waiting from asterisk TO my sip device by default without
having to dial *70 and have asterisk put a CF mark in the database for
my sip
Hi,
Does anyone know... does Sipura have any plans to support GSM or iLBC on any of their devices? Specifically the ATA-2000?
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To
I'm aware that asterisk only supports 20ms packetization rates.
Due to the fact that I will be using some voip devices on a wireless
network which is highly sensative to framerate.. is there any way I can
hard code the packetization rate at say 30 or 40ms and then compile
astrisk? If so, can
IAX is not an option as Sipura devices do not support AIX.Yes, the sipura will handle the different packet sizes...
Is it possible to reprogram asteris to do this?
On Apr 3, 2005 1:55 AM, Steven Critchfield [EMAIL PROTECTED] wrote:On Sat, 2005-04-02 at 21:16 -0500, Matt wrote: I'm aware
* will swallow whatever theSipura sends it.So, I know it's possible to change this in at least onedirection if you are using a Sipura.Bruce KomitoHigh Sierra Networks, Inc.www.servers-r-us.com(775) 236-5815On Sun, 3 Apr 2005, Matt wrote: IAX is not an option as Sipura devices do not support AIX. Yes
Never mind... blah spoke before I thought :P
Found the setting
On Apr 3, 2005 5:23 PM, Matt [EMAIL PROTECTED] wrote:Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue
(audio going from the phone over wireless is slightly choppy).. while
audio coming in (20ms) is ok
, 2005 5:25 PM, Matt [EMAIL PROTECTED] wrote:Never mind... blah spoke before I thought :P
Found the setting
On Apr 3, 2005 5:23 PM, Matt [EMAIL PROTECTED] wrote:Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue
(audio going from the phone over wireless is slightly choppy
Hi,
I'm currently routing my asterisk server out over broadvoice.. it
seems I can do multiple outgoing and incoming calls does anyone
know if broadvoice actually allows this or not?
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]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Monday, April 04, 2005 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] broadvoice
Hi,
I'm currently routing my asterisk server out over broadvoice.. it seems I
can do multiple outgoing and incoming
.
--Dalon
On Apr 4, 2005 11:29 AM, Matt [EMAIL PROTECTED] wrote:
Hi,
I'm currently routing my asterisk server out over broadvoice.. it
seems I can do multiple outgoing and incoming calls does anyone
know if broadvoice actually allows
You probably made the same mistake I did.. you need to use your SIP
password.. NOT your account center password.. log into the account
center... and get your SIP password.. or call them up (apparently they
are more then happy to give it to you if you can provide your phone
number K! They
Well, at the moment I've only done 3... I dunno.. and I don't expect
to have more then that... but who knows?
On Apr 4, 2005 2:41 PM, JD Austin [EMAIL PROTECTED] wrote:
Im curious about that too.. if so how many concurrent calls will they allow?
JD
Matt wrote:
Hi,
I'm currently routing
many concurrent calls will they
allow?
JD
Matt wrote:
Hi,
I'm currently routing my asterisk server out over broadvoice.. it
seems I can do multiple outgoing and incoming calls does anyone
know if broadvoice actually allows
I would say if you want true redundancy.. rsync your files on each
system.. have RAID on each system, and use a load balancing switch.
Incidentally... [EMAIL PROTECTED] Nortel? (eek did I just utter a bad word?)
On Apr 5, 2005 7:53 AM, Infocus [EMAIL PROTECTED] wrote:
I am looking to
mmm one other thought.. the load balancing switch needs to support A)
failover and B) stream assocciation (so that it keeps you on the same
* server as long as you are sending packets!)
On Apr 5, 2005 8:25 AM, Matt [EMAIL PROTECTED] wrote:
I would say if you want true redundancy.. rsync your
Hi,
This is not entirely an asterisk question but I figure someone
here may know the answer to this question.
On several occassions we will lose the ability to use one of our
PRI lines well for our phone system anyway (we also sometimes
lose PRIs on some of our access equipment,
Absolutely! We're using it as such now. If you'd like more
information, e-mail me off list and we could arrange a time to talk.
On Apr 5, 2005 12:44 PM, Video Dery / Internet du Royaume
[EMAIL PROTECTED] wrote:
Hi
Do you think that asterisk could be use as a gateway for residential ip
Hi,
What ports do I need open on the asterisk server (using an iptables
firewall) to allow my sip phones to still work correctly?
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To
I'll elaborate slightly more... the wiki says:
# SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well
iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT
# IAX2- the IAX protocol
iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT
# IAX - most have switched to IAX v2, or
I have a STUN server running on my Asterisk box which seems to work
for most of my SIP clients.. but some of them seem to require NAT=yes
turned on. If I go further and turn QUALIFY=yes to on, is there a
reason I need to keep running a STUN server? If so, what's the
difference?
Hi,
Is there anyway to manipulate the asterisk internal database from the
manager (the one you can telnet to)? And if so.. how does one do it?
(ie for enabling call forwarding, etc)
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Ahh dbput probably will do what I am looking for.. thanks!
On Apr 11, 2005 2:23 PM, Brian Roy [EMAIL PROTECTED] wrote:
On Apr 11, 2005 10:16 AM, Matt [EMAIL PROTECTED] wrote:
Hi,
Is there anyway to manipulate the asterisk internal database from the
manager (the one you can telnet
If the asterisk internal database becomes corrupt... how does one dump
it and start the database over?
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How many people (or remote sip clients) have people actually
seen/gotten to work in a real world environment?
Say a 2.8Ghz machine with a GIG of ram. How many G711 or G729
calls could you handle?
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Hi,
I have the asterisk mysql CDR module/patch installed. But I believe
that's slightly irrelivant to the question but is included for
completeness.
How can I determine from a CDR record (csv by default.. or the mysql
CDR)... if a call was in-network that is.. from one phone to
another?
Steve Beaumont wrote:
Please enlighten me I guess CVS-HEAD is the development version and
CVS-v1-0 satble version ?
Correct.
Don't forget to bookmark http://www.voip-info.org
--
Cheers,
Matt Riddell
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Steve Beaumont wrote:
Thanks, so I download with -r head.
If you want to get the head version, you do not need the -r tag.
--
Cheers,
Matt Riddell
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in other place, both of them
are using G729 for IAX conection.
I wouldn't have thought that you could reliably transmit fax over a
compressed codec. Try ALAW or ULAW (g711u/a).
--
Cheers,
Matt Riddell
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/2.4.22-1.2115.nptl/misc/wcfxo.o: insmod wcfxo failed
What type of card is it?
It is an X100P? If not, you should be modprobing wcfxs or wctdm for
recent versions (with a TDM400P card).
Also, what does dmesg say?
--
Cheers,
Matt Riddell
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can chase down libidn but I
find it odd that others on the list have seemingly gotten asterisk to work
on FC3 but never complained about this particular problem...
Heh, yeah I did too. I ended up just commenting it out of the
asterisk/apps/Makefile.
If you don't need it...
--
Cheers,
Matt Riddell
)
Bearing in mind that the extensions are = extension, priority,
something to do, you seem to be missing s,1...
--
Cheers,
Matt Riddell
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? When loading? Anything strange
about the box? Running any funky firewalls?
--
Cheers,
Matt Riddell
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Hi,
Does anyone know if it's possible to hook an asterisk PBX up to skype?
And if so, any config examples?
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Never mind.. answered my own question looks like their is a bounty
on the ability to do this :P
On Sat, 12 Mar 2005 18:28:00 -0500, Matt [EMAIL PROTECTED] wrote:
Hi,
Does anyone know if it's possible to hook an asterisk PBX up to skype?
And if so, any config examples
Hi,
I've read the wiki... but would like some input from users here (not
implying that wiki writers aren't users).
I'm looking for a cheap (sub 60$) wired phone, or ATA device.. can
anyone recommend one (or several), and possibly a source?
___
Yes...myself. I can be contacted at the email above or on (021) 1387245.
Kind regards,
Matt Riddell
Are there any New Zealand Asterisk users/contractors out there - we're
looking to install a small business pnx and are interested in Asterisk
as a solution
Hi,
Any queries regarding Asterisk in New Zealand should be forwarded to myself.
I can be contacted at the email address above or:
Phone (03) 470 1641 x 818
Cell (021) 138 7245
Fax (03) 470 1645
Can anyone point me in the direction of a Asterisk developer in New
Zealand that we could
I have perl scripts for doing voice contract recording via an extension
including 5 digit codes for each one and the ability to play them back.
Please mail me off list if you are keen.
Hi,
Does anybody know if it is possible to record a conversation with
asterisk ?
Regards
For starters voicepulse is down again at the moment.
matt
Daniel Bichara wrote:
Hi,
I am call Japan via Voicepulse. My IAX Connection to Voicepulse was
sucessfull. But when I put a call (dial), I get an error message:
Feb 19 22:12:23 WARNING[81926]: chan_iax2.c:1128 attempt_transmit: Max
I usually use
[EMAIL PROTECTED]
they do eventually get back to you.
We operate a call centre and have offered them an inbound package, but
it seems they are not interested.
Matt
P.S. Our DID line hasn't been working for around a month nowin the
process of signing up with other companies
an account.
I'm still setting it up, but I'll let you know how it goes.
Matt
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Matthew B Marlowe wrote:
My VP has been up all day without any problems.
Strange...our's wenty down an hour ago and I went onto #asterisk to see
if anyone else was having problems and their service is down also...
are you using gw5.voicepulse.com?
Matt
Ours is back up again now...in the hour it was down we had all staff on
extended lunch break and I signed up with two new providers.
I wonder why you get special treatment?
:-)
Matt
Matthew B Marlowe wrote:
Yes, I am.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
) 255.255.255.255 4569 OK (19ms)
pbx*CLI
Mine:
pabx*CLI iax2 show peers
Name/UsernameHost Mask Port Status
voicepulse 66.234.228.132 (S) 255.255.255.255 4569 Unmonitored
matt/matt(Unspecified) (D) 255.255.255.255 0
, 2004 10:55 AM
Subject: RE: [Asterisk-Users] Can You Specify Codec Per Extension?
Matt wrote:
Hello All,
I was wondering If you can specify which voice codec is used per
extension. I'm using sip phones that support gsm, and some H.323
Endpoints that support GSM, and a couple that don't
chan_oh323. With them both running I can make a call using
chan_oh323 with no errors. Very strange.
Thanks
-Matt
extensions use gsm, while Y extensions use G.711
Thanks -Matt
The Mediatrix Gateways work with Asterisk, however, no gsm support.
Thanks
-Matt
TelCom Products International
2901 Frontage Road S Hwy 10E
Moorhead, MN 56560
Phone# 218-422-9004
Fax# 218-422-9014
Support on MSN Messenger [EMAIL PROTECTED]
- Original Message -
From: Scott Weis [EMAIL
Hello All,
I was wondering if anyone is successfully running asterisk on a system
with solid state storage, such as a compact flash card? I'm looking for some
pointers on doing this.
Thanks
-Matt
___
Asterisk-Users mailing list
[EMAIL PROTECTED
I'm curious what distro of linux you used. I also can't seem to find a
listing of dependancies asterisk requires, even though they are probably
staring me in the face.
Thanks
-Matt
- Original Message -
From: Ernest W. Lessenger [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March
Hello John,
I saw the wiki page on trustix, it said 296 megabytes, still a little
big. I'm downloading trustix now to check it out though.
Thanks
-Matt
- Original Message -
From: John Bittner [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 01, 2004 10:32 AM
Subject: RE
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