To Members,
I currently looking for someone who can configure Cisco IAD's Ie.. IAD 2431
And yes we are more than willing to pay for the service.
If interested please drop me an email
m...@openaccessinc.commailto:m...@openaccessinc.com
Michael DiMartino | Director of IT | Open Access, Inc.
115
To Members,
I currently looking for someone who can configure Cisco IAD's Ie.. IAD 2431
And yes we are more than willing to pay for the service.
If interested please drop me an email
m...@openaccessinc.commailto:m...@openaccessinc.com
Michael DiMartino | Director of IT | Open Access, Inc.
115
To All;
My current issues is a 5 second delay for call that is being transferred
from the Norstar units to
the Asterisk servers VIA a PRI. Is their anything that can be done to
speed up the transfer on the Norstar. Below is my current phone
config.
Norstar1 PRI Asterisk-1
2005 10:15, Michael Di Martino wrote:
My current issues is a 5 second delay for call that is being
transferred
from the Norstar units to
the Asterisk servers VIA a PRI. Is their anything that can be done to
speed up the transfer on the Norstar. Below is my current phone
config.
You need
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Monday, June 27, 2005 5:26 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread
On Monday 27 June 2005 15:46, steve szmidt
I am attempting to get an iaxy device to connect to my
asterisk box over the public cloud however
It fails register and I cannot figure out why.
Below is my iax.conf, iaxy setup file and debug output
from iax2 debug.
My iax.conf
[u7402]
type=friend
accountcode=iaxy
host=dynamic
I am attempting to get an iaxy
device to connect to my asterisk box over the public cloud
however
It fails register and I cannot
figure out why.
Below is my iax.conf, iaxy
setup file and debug output from iax2 debug.
My
iax.conf
[u7402]
type=friend
accountcode=iaxy
host=dynamic
If this list spent at least half the time on helping other asterisk
admins as it does on
trivial things like LiveVoips bankruptcy it just might be a great list.
As it stands now this list is kind of useless. Most request for
assistance with asterisk problems go unresolved of unanswered.
If you
the same
subnet as *?
Michael Di Martino wrote:
I am attempting to get an iaxy device to connect to my asterisk box
over the public cloud however
It fails register and I cannot figure out why.
Below is my iax.conf, iaxy setup file and debug output from iax2
debug.
My iax.conf
at 14:31 -0400, Michael Di Martino wrote:
If this list spent at least half the time on helping other asterisk
admins as it does on trivial things like LiveVoips bankruptcy it just
might be a great list.
As it stands now this list is kind of useless. Most request for
assistance with asterisk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Monday, June 27, 2005 3:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt
On Monday 27 June 2005 14:31, Michael Di Martino wrote
] On Behalf Of steve
szmidt
Sent: Monday, June 27, 2005 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread
On Monday 27 June 2005 14:31, Michael Di Martino wrote:
If this list spent at least half the time
-Original Message-
From: Altus Snyman [mailto:[EMAIL PROTECTED]
Sent: Monday, May 16, 2005 9:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 2 servers via PRI
Good day all
How do i set a connection between 2 asterisk servers via PRI
In Bri I
Hey pooch are u ever going to put up the howto's from the Atlanta
asterisk conference? You only said you would. Don't be like LiveVOIP and
follow thru on your word.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Sunday, June 26, 2005
I am trying to get an iaxy device to connect to my asterisk box
over the public cloud however
It fails register and I cannot figure out why. Below is my
iax.conf, iaxy setup file and out from iax2 debug.
My iax.conf
[u7403]
type=friend
accountcode=iaxy
host=dynamic
secret=u7403p
I am attempting to get an iaxy device to connect to my
asterisk box over the public cloud however
It fails register and I cannot figure out why.
Below is my iax.conf, iaxy setup file and debug output from
iax2 debug.
My iax.conf
[u7402]
type=friend
accountcode=iaxy
host=dynamic
,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Seth
Remington
Sent: Friday, June 24, 2005 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voicemail
On Thu, 2005-06-23 at 23:19 -0400, Michael Di
I am trying to setup voicemail for my iaxy device, however,
i cannot get it to work voicemail never picks up. Below is my config.
Am i doing anything wrong here
>From my Extensions.conf file
exten = 7403,1,Dial(IAX2/7403/10)
exten = 7403,2,Voicemail(u7403)
exten =
Out of the blue i started receiving the following error on my PRI line
which connects my asterisk server to a Norstar 0x32 key system.
The asterisk zaptel.conf file was configure as follows and this config
worked for 6 months until friday. Nothing was changed on either system
prior to friday.
Out of the blue i started receiving the following error on my PRI line
which connects my asterisk server to a Norstar 0x32 key system.
The asterisk zaptel.conf file was configure as follows and this config
worked for 6 months until friday. Nothing was changed on either system
prior to friday.
On Sun, 2005-06-12 at 20:09 -0400, Andrew Kohlsmith wrote:
On Sunday 12 June 2005 06:10, Michael Di Martino wrote:
Out of the blue i started receiving the following error on my PRI line
which connects my asterisk server to a Norstar 0x32 key system.
Well first off, it's likely
On Sun, 2005-06-12 at 20:09 -0400, Andrew Kohlsmith wrote:
On Sunday 12 June 2005 06:10, Michael Di Martino wrote:
Out of the blue i started receiving the following error on my PRI line
which connects my asterisk server to a Norstar 0x32 key system.
Well first off, it's likely
I recently
updated my sip.conf and extensions.conf files and after
shutting
down asterisk and restarting it (asterisk -cvvv)
it shows and
empty dialplan (show dialplan)
*CLI
show dialplan-= 0 extensions (0 priorities) in 0 contexts.
=-
What could
cause somthing like this
below is a
Does
Asterisk support Fedora Core-2 (2.6 Kernel)?
Regards,
Michael
DiMartino Director of MIS
The telx Group,
Inc. 17
State St, 33rd Floor New York, NY 10004
T: 212.480.3300
X2022 C:
646.207.6603
___
Asterisk-Users mailing list
I am
attempting my fist asterisk install.
I do not
have any digum cards so use ztdummy.
However
according to the Asterisk install guide it says to look for one of the
following
USB controller chips.
UHCI USB
controller of OHCI USB Controller.
However,
when I run lsmod to determine
From: Michael Di
MartinoSent: Friday,
April 15, 2005 3:15 PMTo:
asterisk-users@lists.digium.comSubject:
[Asterisk-Users] new install
I am
attempting my fist asterisk install.
I do
not have any digum cards so use ztdummy.
However according to the Asterisk
install guide it says to look
I am
attempting my fist asterisk install.
I do not
have any digum cards so use ztdummy.
However
according to the Asterisk install guide it says to look for one of the
following
USB controller chips.
UHCI USB
controller of OHCI USB Controller.
However,
when I run lsmod to determine
compiled a custom kernel, you may have not
installed/loaded a usb module. If the system is single cpu, you can use a rtc
replacement with zaptel extensions or you can buy even a lowly fxo card. Either
will supply a clock.
From: Michael
Di Martino [mailto:[EMAIL PROTECTED] Sent: Friday, April 15
I have just
inherited a Asterisk box which is configured as follows.
10 internal
Sip Phones
3Pots
Lines
1 voip
provider (SIP)
Call come in
over the pots lines however Outbound goes out thru the VOIP
provider.
However
looking at the configs I cannot figure out what controls how call
Title: Asterisk PBX Manager
Does anyone on this list have any experience Thirdlane.com's Asterisk PBX Manager?
And if so what do you think of it?
Regards,
Michael DiMartino
Director of MIS
The telx Group, Inc.
17 State St, 33rd Floor
New York, NY 10004
T: 212.480.3300 X2022
C:
What is the unsubscribed address?
Thanks
Michael
-Original Message-
From: Steve Underwood [mailto:[EMAIL PROTECTED]
Sent: Friday, February 18, 2005 11:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Re: quadbri and spandsp
You need to
Title: DTMF Payload Type:
To All
I am using a SNOM 190 w/Asterisk server.
Here is my sip.conf
[7501]
type=friend
context=external
username=7501
callerid=Telx 7501 7501
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
My question is this. With above settings in my sip.conf
To All
I am using a SNOM 190 w/Asterisk server.
Here is my sip.conf
[7501]
type=friend
context=external
username=7501
callerid=Telx 7501 7501
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
My question is this. With above settings in my sip.conf specially
dtmfmode=rfc2833
What should
-
From:
Michael Di Martino
To: asterisk-users@lists.digium.com
Sent: Friday, January 14, 2005 3:53
PM
Subject: [Asterisk-Users] Passing PIN
Numbers
To All If anyone can shed any light on this
it would be greatly appreciated. My phones are unable to enter pins numbers
I have the dtmfmode in sip.conf
set to use rfc 2833
however, when my users have to enter pin numbers to join let say
someone's
conference bridge the pin is received twice.
Any ideas on how to solve
this?
___
Asterisk-Users mailing list
Title: Passing PIN Numbers
To All
If anyone can shed any light on this it would be greatly appreciated.
My phones are unable to enter pins numbers correctly when required by the party they are calling.
For example I was given an outside number to attend conference bridge. After the call
Regards,
Michael Di Martino
Director of MIS
The telx Group
Office: 212 480 3300 X.2022
Cell: 646 207 6603
[EMAIL PROTECTED]
--
Sent from my BlackBerry Wireless Handheld
___
Asterisk-Users mailing list
Asterisk-Users
sip phones
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Title: GUI
I am looking for a good Asterisk GUI to manage my server. Any Suggestions?
Regards,
Michael DiMartino
Director of MIS
The telx Group, Inc.
17 State St, 33rd Floor
New York, NY 10004
T: 212.480.3300 X2022
C: 646.207.6603
What is the general consensus on the Polycom SIP Phones?
I am getting random gargled up sounds on mine and I really do think it is the Polycom
Regards,
Michael DiMartino
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I am using the Nortel M7310
I have my Asterisk connected to my Norstar 0x32 MIC software version
4.1 VIA a PRI. Works great
-Original Message-
From: Julio Arruda [mailto:[EMAIL PROTECTED]
Sent: Monday, October 25, 2004 1:22 PM
To: Asterisk Users Mailing List - Non-Commercial
Title: SIP phones
I am looking for a loud ringing SIP phone. I am presently using the Polycom and just cannot loud enough to hear it over the din in a collocation room.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I have the following setup a Norstar MICS 0X32 with 8 POTS Lines
connected to the PSTN, and one ASTERISK server connected to the Norstar
MICS VIA a PRI line.
Now here is the problem I cannot get the MICS to accept a call from the
ASTERISK SERVER when that call is for an outside line(meaning dial
Title: Asterisk and Norstar 0X32 MICS
I have the following setup a Norstar MICS 0X32 with 8 POTS Lines connected to the PSTN, and one ASTERISK server connected to the Norstar MICS VIA a PRI line.
Now here is the problem I cannot get the MICS to accept a call from the ASTERISK SERVER when
to start with i am new to asterisks and i am also a telcom idiot.
with that said i have one vonage line i would like to hook up in my soon to be built
Asterisk ippbx server.
Now with the one Vonage (with call waiting) line can i receive more one call using an
auto attendant route the call the
if I want the auto attendant handle
multilple calls.
For example a call comes in and the auto attendant sends the call to ext 1. Now while
the person on ext 1
Is still conversating can another call be handled by the auto attendant?
Regards,
Michael Di Martino
Director of MIS
The Telx Group
Office
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