thanks
Cassius Smith
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asterisk-users
Hi ALL,
still having trouble getting Ast 12 to run. I got it compiled and built but
now when I try to run, I'm getting a missing library error that seems to be in
error (see below). The .so file DOES exist with correct permissions.
[root@Asterisk12 ~]# asterisk -rvvv
asterisk: error while
On Fri, Oct 18, 2013 at 03:16:08PM -0400, Cassius Smith wrote:
Hello,
I'm trying to build Asterisk12 on a Centos 6.4 VM. The configure script is
erring out with:
…
checking for uuid_generate_random in -luuid... no
checking for uuid_generate_random in -le2fs-uuid... no
checking
(this typically means the uuid
development package is missing)
I have installed (using yum) uuid, uuidd and uuid-devel. No joy, still getting
same error.
Anyone else run into this? How did you get around it?
cheers,
Cassius Smith
On 10/10/11 10:40 AM, Josh Freeman cpe.jfree...@gmail.com wrote:
Hello,
I'm looking at a scenario in which, to make it work, I'd need to dial
into a remote conference from within a local MeetMe room. That might
include being able to dial a conference code after the call to the
remote system
Agree -- make sure you are at the latest firmware.
ALSO: If you have provisioning enabled, and have a duplicate line in your
xml files, that will cause a reboot.
Cheers,
Cassius Smith
On 8/15/11 1:46 PM, C F shma...@gmail.com wrote:
I have 3 Linksys/Cisco 504G phones they keep restarting
Hello all,
I am struggling with an annoying problem. I have an installation with a
small number of Grandstream GXP2010 endpoints. Each endpoint has all the
extensions programmed into the phone for BLF - for instant pickup, transfer
or speed dial.
Except for the Receptionist phone, which is
top posting on purpose
I neglected to say all the extensions can be picked up remotely by the
other endpoints, EXCEPT the receptionist phone x3100. When calls go to that
station, they cannot be picked up. Sorry for the necessity to post twice.
/top posting on purpose
From: Cassius Smith cass
What do you mean by customers? Are you looking for testimonials from
satisfied users?
--
On 7/10/11 11:53 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
How can I find a references customers that used Asterisk as IP Telephony
or Call Center or IVR? In which link they are mentioned?
, so
I'm pretty flummoxed by this
Cassius Smith
[meet-me]
exten = s,1(top),NoOp()
same = n,Answer()
same = n,Wait(1.0)
same =
n,Background(enter-conf-call-numberdigits/0digits/0throughdigits/0digit
s/9)
same = n,WaitExten(5)
exten = 00,n,MeetMe(SouthAfrica0,dMs)
exten = 01,n,MeetMe(Swaziland1
On 7/6/11 3:20 PM, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Cassius Smith
Sent: Wednesday, July 06, 2011 4:37 AM
To: Asterisk Users Mailing List - Non
Hello,
I do not use the skinny firmware. By the way, questions like this are best
shared with the asterisk-users group mailing list, so that a large segment
of the Asterisk community can benefit from the questions and answers.
Cassius Smith
--
On 6/16/11 4:59 AM, bilal ghayyad bilmar
Hello all,
I'm having a problem with my intercom function that I use for under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's
for our general phones. I have a global defined which has all the SIP
channels concatenated together - this is ${ALL-PAGE-EXTS}.
The
On 6/14/11 4:37 PM, Russ Meyerriecks rmeyerrie...@digium.com wrote:
On 6/14/11 4:25 PM, Russ Meyerriecks wrote:
On 6/14/11 9:26 AM, Cassius Smith wrote:
Hello all,
I'm having a problem with my intercom function that I use for
under-chin
paging. I'm running 1.6.2.13 on this server, and we use
2011 16:30, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Cassius Smith
Sent: Friday, May 06, 2011 11:23 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi all
Usually I build asterisk from source, but recently have been doing a
couple of test installations with packages from the Digium repository.
About how long does it take to get from new release announcement into the
Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the
rpm
fixed in svn
On 6 May 2011 16:45, Steve Davies davies...@gmail.com wrote:
On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Cassius Smith
On 5/9/11 6:02 AM, Doug Lytle supp...@drdos.info wrote:
Sebastian Arcus wrote:
Cisco phones (at least the 7940) are supposed to be run with a tftp
server available at all time
That is my experience. But, if you're running tftp under Linux, then
it's probably spawned by xinetd and won't be
Hi all,
I have a production server running with about 90 Cisco 79[46]1's and SIP
release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and
upgraded last night after hours. (Seemed low risk to me!)
Much to my surprise, not a single one of the Cisco 79XX phones would
register. Since it's
,Goto(2,1)
exten = s,n(wrong),Playback(something-terribly-wrong)
exten = s,n,Playback(goodbye)
exten = s,n,Hangup()
Hopefully this is enough to get you started.
Cassius Smith
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The X1 card should seat in the X4 or X8 slots. Check out:
http://computer.howstuffworks.com/pci-express1.htm
HTH
Cassius Smith
On 2/26/11 4:33 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
My server and its slots written in it the following so I need to know
which card to order it (I
On 2/18/11 5:18 PM, Paul Belanger pabelan...@digium.com wrote:
On 11-02-18 03:59 PM, Cassius Smith wrote:
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't get any
ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as
And when the Austria team gets to the office in the morning they will test
it.
(BTW changed TIMEOUT(digits) to TIMEOUT(digit)).
Cassius
On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote:
Hello,
I have an installation in Austria; ISDN service provided by Austria
Telekom
calls and must then transfer.
Is this a p2p vs p2mp issue?
Thanks in advance,
Cassius Smith
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Hi all,
I have a system installation in Guam with two trunks. One has a DSL service
riding on it with the usual filter. That channel however keeps throwing
alarms. I bypassed the filter and it stopped throwing alarms, but of course
the high frequencies annoy the users. I swapped the filters and
CallFwd should be one of the soft keys on your Cisco phones. Are you
re-flashing the Cisco phones with SIP?
-Cassius
On 1/2/11 3:50 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
How to configure the buttons in the Cisco IP Phones to be used for
different functionalities like Call
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith
Sent: Wednesday, November 24, 2010 5:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SPA942 on speaker phone does not hang up?
Hello all,
I am using Linksys SPA942 in my current installation
Premature reply. It did fix the first issue. Now when I ring that phone I
get busy here from the phone, and the call goes straight to voicemail per
dialplan. Maybe another parameter in addition to Reorder Delay?
From: Cassius Smith cass...@cassius.org
Date: Thu, 25 Nov 2010 10:34:25 +0100
Hello all,
I am using Linksys SPA942 in my current installation activity. I see a
peculiar behavior: A call is made and the SPA942 uses its speaker. When the
far end of a call hangs up , the SPA942 stays off hook, and after a time
plays a fast busy. The user then has to press the line presence
Post the germane portions of your xml. How does your phone register each
line button?
Cassius
From: Peter Kowalski kowalla...@gmail.com
Organization: GreatValueMart
Reply-To: kowalla...@gmail.com, Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Date:
I have done something similar; I am using SIP load 8.5.2. I use port 5060 on
both line buttons.
Cassius
From: Peter Kowalski kowalla...@gmail.com
Organization: GreatValueMart
Reply-To: kowalla...@gmail.com
Date: Mon, 22 Nov 2010 13:24:41 -0600
To: Cassius Smith cass...@cassius.org
Cc
.
I'm sure /someone/ has done something like this. I'd appreciate any ideas.
Cassius Smith
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Thanks to all for these replies. I appreciate the variety and this is a
great example of the community supporting one another. I sent this in last
night and awoke to a broad set of replies!
Thanks all - I will post again once I decide on a solution.
Cassius Smith
On 11/15/10 9:09 PM, Sherwood
Hello all,I'm working with one of our offices (that is moving soon) and they're being offered ISDN-10/20/30 services from their TELCO. I'm wondering what kind of interface card I will need (I prefer using Digium's cards). Are the TE121/122/ or TE212/220 series cards compatible with this kind of
BTW I apologize for the double send.
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I'm having trouble getting an IAX2 connection between a couple of servers. Ican make calls from server B to server A, but when I call from Server A to serverB, I get "No authority found".On ServerA I am running Asterisk 1.6.2.9On ServerB I'm running 1.6.2.13Any hints for me? The registrations in
needing to pick up.I apologize for not being able to try this out myself - I'm out of the country with no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated!Regards,Cassius Smith
ith no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated!Regards,Cassius Smith--
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to advertise their
free (!) entry points for Switchvox and FFA. Asterisk training
support - I have no problem with those either. The support and
training are pay-for products, but are a big help to the community also.
My $0.02.
Cassius Smith
Clearly, if Word cannot explain the anguish in his heart,
Mr. Fugina should be using OpenOffice!
Cheers.
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Steve
I have 64 channels being monitored with an SPA962 with two SPA932
sidecars. It works perfectly with Asterisk 1.6.2.9; my users are very
happy with this. Latest firmware is a must.
HTH
Cassius Smith
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* -Original Message-
* From: Todd Reese trees...@gmail.com
* Reply-to: Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
* To: asterisk-users@lists.digium.com
* Subject: [asterisk-users] Dahdi install gone wrong
Sorry for the delay - I lost this message in the middle of a digest.
I tried Answer(2000) and was getting an annoying warning:
[Aug 15 17:20:11] WARNING[15516]: channel.c:1044 __ast_queue_frame:
Exceptionally long voice queue length queuing to DAHDI/1-1
So I changed it back to Wait(2).
I'll try
).
Cassius
-Original Message-
From: Cassius Smith cass...@cassius.org
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users
Hi all,
There are a lot of posts around the web about my question; unfortunately
I have not been able to get any of the solutions to work. I'm using
Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working
for the secretaries that monitor their bosses' phones.
The BLF and the
, are the extensions set to be in the same
pickupgroup?
regards
ron
On 8/15/10 7:01 AM, Cassius Smith wrote:
Hi all,
There are a lot of posts around the web about my question; unfortunately
I have not been able to get any of the solutions to work. I'm using
Asterisk 1.6.2.8 under CentOS 5.5. I'm trying
Hi list,
I'm having a problem with CallerID names not showing up when calls come
in. I have dialplan code to store the callerid(name) away and it is
blank (null). However, the voicemail variable ${VM_CALLERID} has the
name field populated. For example, here is some of the dialplan code:
2.
Thanks Warren. That fixed it.
I am using T1's and didn't think the spill would take that long.
Ciao,
Cassius
Add a Wait(2) before your first Set statement. Sometimes callerid
takes a
few seconds to arrive over the line, depending on your technology.
--
The above entry works, but:
[SPIDR-3749](caryspider)
mailbox=3...@default
This one doesn't.
[caryspider] looks like this:
[caryspider](!)
type=friend
context=users
host=dynamic
secret=xx
Any ideas? I'm stumped.
Cassius Smith
other participant endpoint into the conference, but no more.
I know I can (and will) use MeetMe to do large conferences. My
question is - am I forced to do so by SIP? Or am I missing something?
Thanks!
Cassius Smith
for
the mailbox.
I am flummoxed. Any ideas welcome!
Cassius Smith
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OK, feeling very stupid right now.
The test mailbox had delete=yes option set. All cleared up; sorry for
cluttering up the list.
Cassius
snip
Now, however, I don't get message waiting lamp to show up on the phones
and when the recipient of a voicemail tries to retrieve the message
Alyson says
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