thanks
Cassius Smith
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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 loadi
>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
(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
On 10/10/11 10:40 AM, "Josh Freeman" 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 was answered.
>
>
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" wrote:
>I have 3 Linksys/Cisco 504G phones they keep restarting
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.
From: Cassius Smith
Date: Fri, 05 Aug 2011 15:31:14 -0500
To
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 handle
What do you mean by customers? Are you looking for testimonials from
satisfied users?
--
On 7/10/11 11:53 AM, "bilal ghayyad" 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?
>
>Regards
>Bi
On 7/6/11 3:20 PM, "Eric Wieling" 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:
other installations, 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-number&digits/0&digits/0&through&digits/0&digit
s/9)
same => n,WaitEx
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&qu
On 6/14/11 4:37 PM, "Russ Meyerriecks" 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
>>
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 ${ALL-PAGE-EXTS}.
The problem
gt;> https://issues.asterisk.org/view.php?id=18951
>>>
>>> fixed in svn
>>>
>>> On 6 May 2011 16:45, Steve Davies wrote:
>>>> > On 6 May 2011 16:30, Eric Wieling wrote:
>>>>>> >>> -Original Message
Hi all
Usually I build asterisk from source, but recently have been doing a
couple of test installations with packages from the Digium repository.
About how long does it take to get from new release announcement into the
Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the
rpm
t;> https://issues.asterisk.org/view.php?id=18951
>>
>> fixed in svn
>>
>> On 6 May 2011 16:45, Steve Davies wrote:
>>> > On 6 May 2011 16:30, Eric Wieling wrote:
>>>>> >>> -Original Message-
>>>>> >>>
On 5/9/11 6:02 AM, "Doug Lytle" wrote:
>Sebastian Arcus wrote:
>> Cisco phones (at least the 7940) are supposed to be run with a tftp
>> server available at all time
>
>That is my experience. But, if you're running tftp under Linux, then
>it's probably spawned by xinetd and won't be running un
Hi all,
I have a production server running with about 90 Cisco 79[46]1's and SIP
release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and
upgraded last night after hours. (Seemed low risk to me!)
Much to my surprise, not a single one of the Cisco 79XX phones would
register. Since it's
=> s,n(daycheck),GotoIfTime(08:00-16:59,mon-fri,*,*?open)
exten => s,n,Set(MENU=night-menu)
exten => s,n,Goto(night)
exten => s,n(open),Set(MENU=day-menu)
exten => s,n(night),NoOp()
exten => s,n(top),Wait(0.5)
exten => s,n,GotoIf($[${COUNTER}>=10]?wrong)
exten => s,n(p
iles being 0 bytes. I.e. Touch the tlv files but
> leave them empty.
>
> HTH
> 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" wrote:
>Hi All;
>
>My server and its slots written in it the following so I need to know
>which card to o
On 2/18/11 5:18 PM, "Paul Belanger" 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 no
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 expecte
any thanks for this idea, Christian I have put this equivalent into the
dialplan
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 wrote:
>> Hello,
>>
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 the
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" wrote:
>Hi All;
>
>How to configure the buttons in the Cisco IP Phones to be used for
>different functionalities like "Call Forward, Call Pickup
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
Date: Thu, 25 Nov 2010 10:34:25 +0100
To: Aste
ng up?
It is the phone itself: go to Regional tab and scroll down to Reorder Delay
and make it 255. That tells it not to play re-order tone and just hangup.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith
Sent: Wednesday
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 butt
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
Organization: GreatValueMart
Reply-To:
Date: Mon, 22 Nov 2010 13:24:41 -0600
To: Cassius Smith
Cc: 'Asterisk Users Mailing List - Non-Commercial Discu
Post the germane portions of your xml. How does your phone register each
line button?
Cassius
From: Peter Kowalski
Organization: GreatValueMart
Reply-To: , Asterisk Users Mailing List -
Non-Commercial Discussion
Date: Mon, 22 Nov 2010 12:38:22 -0600
To:
Subject: [asterisk-users] asterisk
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, "She
ERY inexpensive.
I'm sure /someone/ has done something like this. I'd appreciate any ideas.
Cassius Smith
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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 ser
BTW I apologize for the double send.
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as "UNAUTHENTICATED".
On ServerA I am running Asterisk 1.6.2.9
On ServerB I'm running 1.6.2.13
Any hints for me?
The registrations in both directions seem to work fine when I do an iax2
reload from the CLI.
config file snips shown below.
Thanks
Cassius Smith
==
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 bo
ck 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--
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at each phone without the user 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 mos
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 S
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
* Reply-to: Asterisk Users Mailing List - Non-Commercial
Discussion
* To: asterisk-users@lists.digium.com
* Subject: [asterisk-users] Dahdi install gone wrong
* Date: Mon, 23 Aug 2010 10:26:58 -0400
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
context includes).
Cassius
-Original Message-
From: Cassius Smith
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] BLF/Call Pickup using SPA942, SPA962,
SPA932
Date: Sat, 14 Aug 2010
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 to get call pickup working
> for the secreta
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 spee
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.
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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. Set
R-3758](caryspider)
mailbox=3...@default
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=x
pants. I can add one
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?
T
OK, feeling very stupid right now.
The test mailbox had "delete=yes" option set. All cleared up; sorry for
cluttering up the list.
Cassius
>
>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
to the INBOX directory for
the mailbox.
I am flummoxed. Any ideas welcome!
Cassius Smith
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