Hi,
I didn't have this gateway, But on welltech 4fxo gateway. You can just dial
SIP/[EMAIL PROTECTED] Even the gateway didn't register to the
asterisk server.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Corne
Vermeulen
Sent: Monday, January 30
set dtmfmode=rfc2833 in sip.conf and try
again.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
AislingSent: Wednesday, February 01, 2006 11:03 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] XLite dtmf
issue?
Hi,
I’m wondering if anyone has
experienced
Hi,
In AAH, you can setup the "Incoming Calls" to ring your extension. Or to
ring extensions in a ring group.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Smith
Sent: Friday, February 03, 2006 6:53 AM
To: 'Asterisk Users Mailing List - Non-Comme
Hi,
http://www.voip-info.org/wiki/view/Asterisk+sound+files+international
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lang
Sent: Monday, February 06, 2006 3:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] French and Germa
Hi,
Have you try to search on eBay? I found some welltech devices for sale.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph
Sent: Tuesday, February 07, 2006 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asteri
Not sure answer your question? Try to write some html code and let user
register the username & password online.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy
Sent: Tuesday, February 07, 2006 7:31 AM
To: asterisk-users@lists.digium.com
Subj
Hi,
In my remember. The AAH will automatic config the TDM04B
card. try the CLI command "zap show channels"
regards,
kevin
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nelson
GranadosSent: Tuesday, February 07, 2006 7:18 AMTo:
asterisk-users@lists.digium.comSubject:
FYI, I have two welltech SIP products. 3702a (2FXS+2FXO) and 3804 (4FXO).
And all upgrade to the 12/2005 latest firmware. Actually the new release
firmware most of bug fixed. The FXO ports still can't detected the barecore
callerid in Taiwan telcom.
These two voice gateway work fine with the ast
In my remember, when playback a file. The Asterisk will automatically choose
the audio file with the lowest conversion cost. Not always looks the
filename.gsm.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: Thursday, February 09, 2006
FYI. The welltech 3702 (or 3804) latest firmware didn't have sip register
problem on fxo port.
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vahan
Yerkanian
Sent: Tuesday, February 07, 2006 8:11 PM
To: Asterisk Users Mailing List - Non
yes, it's work.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roberto
PereyraSent: Thursday, January 12, 2006 8:15 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Web based SIP client
HiI found this http://www.etntalk.com/callto/l
Hi,
I have success upgrade two 7960 phone from sccp to sip.
Some tftp server doesn't work. You can try this tftp server and post your
tftp logs.
http://www.solarwinds.net/Tools/Free%5FTools/TFTP%5FServer/
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
EdwardsS
FYI,
http://www.cepstral.com/
You can download the english and spanish voice files for test first. And
modify the festival-script.pl to using cepstral swift program.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Otero
Sent: Thursday, January
Hi,
You need the unattended transfer (blind transfer) featuer. That implemented
in Asterisk (#) button. Not attended transfer.
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas Artner
Sent: Friday, February 10, 2006 8:01 AM
To: aster
Hi,
Can you make some test to send voicemail to other mail account? (e.g,
@yahoo.com, @hotmail.com...). If it's work. I think not a SMTP authetication
problem. Or you can check the asterisk maillog first.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Beha
Hi,
I have another model 3702a (2FXS & 2FXO) voice gateway. You can implement
one-stage dialing on this device.
1. using "sip show peers" to make sure two ports (1fxo/1fxs) was registered
to asterisk.
2. login 3701 web and change the defualt routing. The factory default is IP
-> FXS port. Change
In my remember, the artdio ipf-3000 phone support instant message.
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman
Sent: Thursday, February 23, 2006 8:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Have you try first blind transfer to a meetme meeting room. Then multiple
user can join in.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert La
Ferla
Sent: Thursday, December 15, 2005 3:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussio
The hyperlink work now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Takayuki
Uehara
Sent: Friday, December 16, 2005 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Alternative source for Asterisk-IM
I tr
Hi,
Asterisk High Availability Solutions
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
SteveSent: Saturday, December 17, 2005 10:14 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Reden
>> How do I change the time zone for Asterisk? Currently the system time is
correct but when I dial *60 it reports a different time (out by many hours).
In [EMAIL PROTECTED] console type "config" type to change time-zone
___
--Bandwidth and Colocation
It's the welltech wellgate 3804 4FXO gateway.
More info:
http://www.welltech.com/product_e_03.htm
I have another model 3702 (2FXO+2FXS).
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta
Sent: Monday, January 02, 2006 2:09 PM
To: [EMA
http://www.alibaba.com/catalog/10886425/Fxs_fxo_Port_Converter.html
I have one and bad voice quality.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, January 02, 2006 8:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discu
try this:
http://www.asteriskguru.com/audio_conversion.php
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
HarrisonSent: Tuesday, January 03, 2006 10:52 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] offline
g.729 transcoding
I'm trying to get some of t
FYI:
http://redice.krisk.org/
g729:
http://www.readytechnology.co.uk/open/ipp-codecs/
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bogdan
MoldovanSent: Tuesday, January 03, 2006 1:46 PMTo:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE:
[Asteri
Hi,
Draytek 2900 is a great router. Easy to setup & stable. I want known more
detail of your network & configuration. I can setup it and make some test.
Regards,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Attwood
Sent: Saturday, January 0
Now draytek have some SIP embeded router (e.g., 2100VG, 2900VG...). Maybe
you can try these new router.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergio
Chersovani
Sent: Saturday, January 07, 2006 9:16 PM
To: Asterisk Users Mailing List - Non-Comme
Try this.
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antonio Gallo
Sent: Saturday, January 07, 2006 8:20 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] wich IAX soft cl
Thanks a lot. It's work :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Tuesday, January 10, 2006 10:18 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Re: Re: Remotely
Hi,
You mean Cisco "FXS" Port? Can you describe more detail about your network
configuration?
Regards,
Kevin
> _
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Timothy R.
> McKee
> Sent: Thursday, January 12, 2006 6:19 AM
Sometime are phone's configuration change. Because Cisco or Polycom sip
phone download the settings from the tftp server after reboot.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Langstaff
Sent: Wednesday, January 11, 2006 11:47 PM
To: Asterisk U
The company name: WellTech, and the model number: WellGate 3804 (4FXO).
Support H.323 or SIP. You can download these firmware from there site.
http://www.welltech.com.tw
I have one 3804 on my desk now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EM
This device only support FSK (Bellcore) & ETSI callerid. But I have the same
problem and test with there RD now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Sunday, January 01, 2006 2:27 AM
To: Asterisk Users Mailing List - N
Hi,
I download the guide from yoda site. It's seems the original vendor is Accel
AmiGate Elite 400 (http://www.accel.com.tw/frame/frame_age400.htm)
I have one H.323 model and can't upgrade to SIP firmware. So what is your
firmware version?
Regards,
Kevin
-Original Message-
From: [EMAIL
Hi,
To call the extensions registered on Asterisk. You don't need th gatekeeper.
In your H.323 devices just set the "gateway" to Astiersk IP. I have test on
ooh323 channel drive & netmeeting.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Salas
Hi,
Maybe buy 7912 phone and register to CCM is another choice.
or integrated CCM with asterisk voicemail system.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sys
readSent: Tuesday, January 24, 2006 11:28 PMTo: Asterisk
Users Mailing List - Non-Commercial Discussion
Hi,
AAH 2.7 + Digium TE100P work great on our lab. Some example.
file: /etc/zaptel.conf
# Span 2: ZTDUMMY/1 "ZTDUMMY/1 1"
# Global data
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
file: /etc/asterisk/zapata.conf
[channels]
language=en
context=from-pstn
signalling=pri_cpe
switchtype=natio
In my remember. You can't select the port by dial command. The welltech 38xx
voice gateway pick a free port and dial-out.
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Friday, March 24, 2006 5:57 AM
To: asterisk-users@li
Yes,
Same configuration as Martin.
1.for incoming call just set the 3804 "hotline" to one sip
extension number.
2.for outgoing call, you just using regular dail command to
pass the phone number to 3804 (3804 is a 4FXO port device, the call from
ip side always pass to FXO Port). You can tel
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Friday, March 31, 2006 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] welltech Wellgate 3804 in SIP mode
one-stage calling function?
On 3/30/06, kevin
AAH 2.8beta1 include the FreePBX.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Thursday, March 30, 2006 9:44 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: FreePBX & AAH
In article <[EMAIL PROTECTED]>,
Jim
Hi,
I have download the uplink and test with skype 1.4 & 2.0. not lucky to me.
Only connect on first call then hang. I need to reboot my windows xp
everytime.
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon
Sent: Wednesday, Ma
In my remember. The uplink install a virtual sound card. So uplink can auto
answer the call from skype or sip side and redirect to another side. No
matter what kind of onboard audio card do you have.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Pe
Hi
Sorry for my chinese engish first.
The VIP-320 seems like a SIP ATA+DECT Phone product. Please check you have
registered to the asterisk server first. Because the VIP-320 built-in
H.323/SIP dual mode. The web config tool not so clear. Reference the
Page.31. Input the "SIP:asterisk_ip_address"
Hello, ,
I have some of asterisk to CCM4 SIP trunk oneway audio problems.
I have setup a asterisk server. It's work great and have no
any problems connect to local ITSP (using SIP protocol) . But we need to
build a sip trunk to another CCM4 server.
The network typology like this.
SI
Hi,
I have the same problem on TE110P and Taiwan telco PRI line. I think to fine
tune the rigntone frequencies not resolve this problem.
For example.
When I make a call to mobile. I can hear one ringtone like geneate by
asterisk or device. And another ringtone like from telco. You known, some
mo
Hi
Thanks a lot. It's work for my ArtDio IPF-3000 phone. I have make a lot fine
tune on the zapata.conf file. Doesn't have any help.
Just add progressinband=no in the sip.conf. Done!
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
Sent
Hi,
The CCM4 behide the PIX firewall? Have you open the
ports for SIP trunk on CCM4 side? (TCP/UDP 5060, UDP
16348-32768)
Kevin
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antonio
Serrano LuqueSent: Wednesday, March 01, 2006 4:18 PMTo:
asterisk-users@lists.digium.c
Hi,
"Admin Login" > "Ext 1" > "Proxy and Registation /
Proxy"
enter the asterisk ip address here.
Kevin
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damian
FunnellSent: Tuesday, April 11, 2006 8:14 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionCc: Jam
Hi,
It's work on my spa-941. I just add belowing line before dial the extension.
exten => s,3,SIPAddHeader(Call Info: Anwser-After=0) ; This is for the Snoms
and Others
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Tuesday, Ap
Hi,
Our company replace the CCM3 with Asterisk this month :-)
Take a look here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Voic
email+Integration
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terry Wade
Sent: Tuesda
--Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
(port)
Sent: Tuesday, April 11, 2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] App Page() in 1.2.5
kevin ling wrote:
>Hi,
>
>It'
Hi,
I use the AAH2.7 (asterisk version 1.2.5). When someone call me and I pickup
the phone. If I want to transfer to another extension. Then I dial the "#"
key the system will play the onhold music. After I dial the extension
number. The system stop play onhold music and play ringtone. Is it pos
Hi,
Have you try to set hidecallerid=yes in zapata.conf?
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre
Courchesne - Consultant
Sent: Friday, April 14, 2006 12:02 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Display "Con
Hi,
I have try install the TE110P or TDM400P on HP Proliant DL380 Server. And
use the AAH 2.7 distribution. When I reboot the server. The CentOS always
display "some hardware removed - Tiger pci card." & "Found new
hardware - Tiger pci card". Is it the digium cards have some
compatia
Hi,
Check the vm_general.inc file
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Sunday, April 16, 2006 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] voicemail email-from
Ho
Hi,
We setup a HP DL360G2 (Xeon 2.8*2, 2G RAM)server & Digium TE110P (E1 PRI to
telco) for a small office. Include 70 IPF-3000 phone in office & 10 phones
on another warehouse. Between the office and warehouse we use the G.SHDSL in
bridege mode to connect each other. I suggest you can setup a smal
Hi
Check this setting:
bindaddr = 0.0.0.0 :IP Address to bind to (listen on)
kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomislav
Par?ina
Sent: Tuesday, April 18, 2006 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi,
Check the script. You can assign the voice by -n option, e.g.,
/opt/swift/bin/swift -n Diane
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane Young
Sent: Friday, April 21, 2006 9:17 PM
To: Asterisk Users Mailing List - Non-Commercia
Hi,
I only check the AAH & AMP. The inbound routing "from-pstn" didn't include
the context "ext-group". So the ring group setting doesn't work when you
call from PSTN.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo
Sent: Tuesday, April
Yes, just set the hotline number to an extension number. And disable the
welltech IVR function.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Artifex
Maximus
Sent: Tuesday, April 25, 2006 3:51 AM
To: Asterisk Users Mailing List - Non-Commercial D
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giorgio
Incantalupo
Sent: Wednesday, April 26, 2006 4:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Excessive Asterisk delay to answer on ZAP
inboundcall
Hi,
I have an asterisk 1.2.1
Hi,
I have make some test. If asterisk can decode the callerid. The asterisk
will answer the call after 2 rings. But when asterisk have some problem to
get the callerid. Asterisk pickup the call after 3-4 rings.
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PR
Hi,
Same dial pattern on my extension.conf, But it's work great. The Asterisk
only match 7 digits number. My * version is 2.1.6.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent: Thursday, April 27, 2006 8:35 AM
To: Asterisk Users Ma
Hi,
Check the DISA command.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DISA
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jnuoiqweahf
kajhdsff
Sent: Thursday, April 27, 2006 12:21 PM
To: asterisk-users@lists.digium.com
Subject: [
So sorry, the correct version is 1.2.6 :-)
kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Thursday, April 27, 2006 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Re: Pattern ma
On Behalf Of Andrew
Kohlsmith
Sent: Thursday, April 27, 2006 9:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Re: Pattern matching problem
On Wednesday 26 April 2006 20:54, kevin ling wrote:
> Same dial pattern on my extension.conf, But it's work great. The
>
OTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, April 27, 2006 9:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Re: Pattern matching problem
On Wednesday 26 April 2006 20:54, kevin ling wrote:
> Same dial pattern on my extension.conf, But it's work great. The
Hi,
Have you try to install this TDM400P card on another asterisk server? Same
problems?
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shyam Gopale
Sent: Thursday, April 27, 2006 5:07 PM
To: asterisk-users@lists.digium.com
Subject: [As
Hi,
You can call an agi script to convert the text file to wave format.
Example:
http://www.voip-info.org/wiki/view/swift.agi
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Joseph
Sent: Monday, May 01, 2006 7:08 PM
To: Asterisk Users
Subjec
Hi,
What protocol for your 7960 phone? SCCP or SIP? You can turn on the SIP
debug on CLI to make sure the callerid and name pass to your phone.
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Tuesday, May 02, 2006 12:37 AM
To:
Hi,
But it's seems the auto-answer function work on my spa-941. Have you upgrade
to the latest firmware version?
Regards,
kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Friday, May 05, 2006 9:02 AM
To: Asterisk Users Mailing Lis
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