Does anyone know if sip phones from any of the major IP phone vendors
support 802.1x authentication? Any feedback would be greatly
appreciated.
Thanks in advance.
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624-5943
Web
: [asterisk-users] IEEE 802.1x capable sip phones
Jeronimo Romero wrote:
Does anyone know if sip phones from any of the major IP phone vendors
support 802.1x authentication? Any feedback would be greatly
appreciated.
This is so unlikely. I worked on 802.1X and 802.11i. There is just too
much
Running asterisk 1.2.7 with latest zaptel on centos4.4. with Aastra 55i phones.
Local outbound calling works fine, but ATT requires clients enter 7 digit code
for long distance. All calls with 7 digit code are lost within 20 seconds of
the call. This is the message Im getting:
Apr 19
Just be careful with the sidecar. It was to be screwed on and the screws
that come with the unit strip very easily. Make sure you have a nice
electronics grade screwdriver with a long thin shaft or you'll have
trouble with the side car.
Another really nice feature of this phone is that the BLF
Is there any technical difference between a T1 cable and a cat5e patch
cable as far as using them with Digium T1/E1 cards? Can PRI circuits
terminating at a smart jack connect successfully to Digium cards using
straight through CAT5e cables? If so, are they using all of the pins in
the cable?
I assume that I would need to cross these pins over if I were going from
t1 card to t1 card. Is this correct?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Sent: Sunday, March 18, 2007 7:17 PM
To: Asterisk Users Mailing List - Non-Commercial
/Return/Ground
6
6
Shield/Return/Ground
On 3/18/07, Jeronimo Romero [EMAIL PROTECTED] wrote:
I assume that I would need to cross these pins over if I were going
from
t1 card to t1 card. Is this correct?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
It would be cool if you could add some kind of login script capability to nodes
in sip.conf and iax.conf.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: Sunday, March 18, 2007 11:16 PM
To: Asterisk Users Mailing List -
, 2007 12:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] TE110P: Error == Asterisk died with code
1.
On Wed, Feb 28, 2007 at 10:47:48AM -0500, Jeronimo Romero wrote:
Thank you all. Was a signaling issue.
And for the benefit of those who will read the archive: how have
.
On Tue, Feb 27, 2007 at 08:01:41PM -0500, Jeronimo Romero wrote:
Running Asterisk 1.2.9. I just installed a TE110P card and configured
zaptel.conf zapata.conf. The config files look right to me but I'm
getting the following error when trying to start asterisk:
Asterisk died with code 1
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0
txgain=0.0
immediate=yes
group=0
signalling=pri_cpe
context = from-pstn
channel =1-23
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624
server as a voicemail server
forlegacy PBX -- FXO or FXS???
Will the Asterisk box be hooked up to external lines on the Merlin, or
extension lines?
External - FXS
Extension - FXO
later,
PaulH
On Mon, 2007-02-05 at 20:03 -0500, Jeronimo Romero wrote:
Hey All,
I'll be configuring
PROTECTED] On Behalf Of Jeronimo
Romero
Sent: Monday, February 05, 2007 8:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk server as a voicemail server for
legacyPBX -- FXO or FXS???
Hey All,
I'll be configuring an asterisk box to be the voicemail
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo
Romero
Sent: Monday, February 05, 2007 8:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk server as a voicemail server for
legacyPBX -- FXO or FXS???
Hey All,
I'll be configuring
Hey All,
I'll be configuring an asterisk box to be the voicemail server to an old
Merlin system which had an octel 100 voicemail server that is now dying.
My question is simple: do I need to stick an FXO card in the asterisk
box? My logic is that if the Merlin Magix system is actually
We have the 2950. It came with only 2pcix ports. And if you need to
power an fxs card, you need to route wires around. It wasn't easy to
work with.
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
I'm going to be rolling out asterisk at a small office and one requested
feature was the ability to have a phone that can be configured so that
ringtones can be configured according to the callerid of the caller.
Does anyone have Asterisk experience with such a phone? Any suggestions
would be
Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown).
The Oreilly (Asterisk: Future of Telephony) book mentions it in passing
saying that all you need to enable it is to set immediate=yes in
zapata.conf. Has anyone implemented this in brokerage trading
environments?
Thanks
I'm trying to set on-demand call recording. Here's a snippet of the
pertinent dialplan. The purpose of this is to allow one user in
particular to be able to receive an email recording of the call
everytime he dials *91 + number. The problem is that the email is not
going out or being generated
Hey everyone. I recenty installed a server at a datacenter offsite and
the thing is getting hammered with invalid ssh logins so I decided to
use some iptables.
I included my ruleset here. I was wondering if I could get some feedback
based on my ruleset from those of you using iptables in
The problem was that autofallthrough=yes was set in extensions.conf
I'm experiencing a strange problem. My inbound calls are hanging up
right after Background() message even though response timeout is set
to
10 sec.
[voicepulseincoming]
exten=_X.,1,Answer
I'm experiencing a strange problem. My inbound calls are hanging up
right after Background() message even though response timeout is set to
10 sec.
[voicepulseincoming]
exten=_X.,1,Answer
exte=_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1)
I've installed on 6.1 it from ports with ztdummy without an issue. I've
never used zaptel hardware on it though. Had some issues with meetme
and ztdummy but all worked out.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of J. Oquendo
We are doing a medium sized office in NYC with 80 phones. The customer
originally requested Polycom 601 phones. The COO also authorized us to
purchase 2 Grandstream GXP2000 phones for the mail room. We find these
phones much easier to configure and work with asterisk . They support
BLF
We are currently using xten eyebeam soft phones for our laptops at work.
We would like to know if it would be possible to configure the phone to
auto pickup when it reads the sip header:
Call-Info: answer-after=0
Is this possible with this soft-phone or any other soft-phone ?
thanks in
How is your DTMF mode configured in sip.conf
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Troy
Perkins
Sent: Wednesday, November 15, 2006 7:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] simple mainmenu ivr tones not
Thanks for the input. I take it the snoms support both BLF intercom?
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
==
From: [EMAIL PROTECTED
We've had great results with Astrocom powerlink for load balancing
outbound wan connections.
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
==
-Original Message-
From
Sorry if you see this message repeated twice. I would like to set up
hard phones and softphones with the same extension so that when anybody
in the company dials an extension, each user's hardphone and softphone
will ring at the same time. I've tried setting this up before, but I
noticed that the
Of Anselm
Martin Hoffmeister
Sent: Sunday, November 12, 2006 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] same extension on softphones and
hardphones
Am Sonntag, den 12.11.2006, 16:29 -0500 schrieb Jeronimo Romero:
Sorry if you see this message
Im running asterisk 1.2.8. I would like PSTN inbound
calls to do the following:
1-once PSTN callers enter their desired extension; they have
to record their name
2-recording then announces that it is trying to locate the
user
3-asterisk calls local extension and announces callers
,2,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(u${ARG1})
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(b${ARG1})
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
==
Jeronimo Romero
EUS Networks
Email
Im running Asterisk 1.2.8 with Polycom ip501s xten
softphones The only problem Im experiencing is the following: I
cant seem to get blind transfers to work with my Polycom 501 phones Either
through the feature code or the soft keys.
Feature code blind transfers:
I set up a feature
For the soft buttons to work the way you want it, make sure you got
the Polycom dialplan setup right.
On 10/29/06, Jeronimo Romero [EMAIL PROTECTED] wrote:
I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only
problem I'm experiencing is the following: I can't seem to get blind
It was the digit map. Thanks.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo Romero
Sent: Sunday, October 29, 2006 6:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] blind transfers with IP Polycom
- Default timeout (3 seconds)
Any digit followed by a 3 second timeout will match. You can include
pattern to match * and #.
xx.T|*x.T|#x.T
Julian.
On 10/29/06, Jeronimo Romero [EMAIL PROTECTED] wrote:
Do you mean the digitmap??
-Original Message-
From: [EMAIL PROTECTED] [mailto
Has anyone tried RedFone?? It is supposed to offload a lot of that bus
overhead to the external unit doing TDMoE.
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
==
-Original
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH distorted on Pound Key Linux on
asterisk1.2.8
Remove mpg123. In the Asterisk source directory type make mpg123 I
believe that make install is required to install it.
Jeronimo Romero wrote:
Running Asterisk
that extreme.
lets see your conf's and if you made your own music, or default MOH
From: Jeronimo Romero [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MOH distorted
, but it is terribly
loud and mistorted. Tried running under quitemp3 profile but it didnt
help.
I feel like there is
something I may be missing here. Any ideas???
Thanks in advance.
Jeronimo.
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office
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