Displaying 19 results from an estimated 19 matches for "eusnetwork".
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eusnetworks
2006 Oct 29
4
blind transfers with IP Polycom 501
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 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 map in features.conf like this:
blindxfer => #
This works for all my
2006 Nov 09
3
announcing inbound PSTN calls
I'm 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 recorded name
4-local recipient user can choose to take the call, send it to voicemail
or transfer it to
2006 Nov 15
2
Grandstream GXP2000 -- What's the Catch?
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 & intercom right out of the box. They can also be centrally managed
and provisioned. They also sound great
2007 Mar 18
6
T1 cable for Digium T1/E1 Cards
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?
Thanks in advance
2006 Nov 21
2
FW: CISCO 7960G & Asterisk
I was wondering if people have experienced issues with Cisco 7960G and
Asterisk.
Any feedback on people's experience deploying this phone in production
environments would be appreciated.
Thanks in advance
2006 Oct 30
1
dealing with blind transfers to invalid extensions
Running Asterisk 1.2.8 kernel 2.6.13.4-1.
Everything in my dialplan seems to be working well except for one
problem.
When calls are blind transferred to an invalid extension I would like
the call to go to the operator on ext 1000?
What is the best way to do this? Thanks in advance
Here's a snippet of my extensions.conf
[default]
exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
2006 Nov 27
0
calls hang up even after Background() messageeventhough response timeout is set to 10 sec
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
> exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1)
>
2006 Dec 11
2
asterisk PLAR
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 in advance.
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2007 Feb 06
0
asterisk server as a voicemail server forlegacy PBX -- FXO or FXS???
I believe it will be hooked up to extension lines.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Hales
Sent: Monday, February 05, 2007 8:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk server as a voicemail server
forlegacy PBX -- FXO or FXS???
2008 Oct 21
1
Does Asterisk support SIP Join Headers
I'm wondering if the SIP header join, RFC 3911, is supported in the
asterisk stack?
--
*****************************
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
****************************
2007 Apr 19
1
aastra phones with asterisk 1.2.17 - hangup after 20 seconds
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 I?m getting:
Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission
2007 Feb 05
2
asterisk server as a voicemail server for legacy PBX -- FXO or FXS???
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 generating
electrical current, then I would need to have an fxo card. Is this
correct?
2006 Nov 12
2
same extension on softphones and hardphones
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 last sip device to register with the same extension is
the only one that rings when the
2007 Feb 06
1
asterisk server as a voicemail server forlegacyPBX -- FXO or FXS???
Thanks. Is there a way I can log into the Merlin Magix to determine
that? How else do I tell?
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric
Germann
Sent: Monday, February 05, 2007 8:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users]
2006 Nov 27
1
calls hang up even after Background() message eventhough response timeout is set to 10 sec
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)
exten=>_X.,n,GotoIfTime(9:00-15:00|fri|*|*?business-hours,s,1)
exten=>_X.,n,GotoIfTime(*|*|*|*?after-business-hours,s,1)
2006 Dec 08
1
cal recording with email
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 when I use the ${CALLFILENAME} variable.
When I use the actual file name of the gsm recording,
2007 Jan 03
3
caller id ring tones for Asterisk Phone
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 greatly appreciated.
Thanks in advance!!!
2007 Feb 27
5
TE110P: Error ==> Asterisk died with code 1.
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.
Automatically restarting Asterisk.
Does anyone have any idea what is wrong with this configuration??
Thanks in advance!!!
Here's my config files:
zaptel.conf
2008 Jan 10
8
IEEE 802.1x capable sip phones
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: jromero at euscorp.com
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
======================
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