Displaying 12 results from an estimated 12 matches for "cwallace".
Did you mean:
wallace
2008 Mar 03
7
DO NOT REPLY [Bug 5299] New: 2.6.9 client cannot receive files from 3.0.0 server
https://bugzilla.samba.org/show_bug.cgi?id=5299
Summary: 2.6.9 client cannot receive files from 3.0.0 server
Product: rsync
Version: 3.0.0
Platform: x86
OS/Version: Windows XP
Status: NEW
Severity: major
Priority: P3
Component: core
AssignedTo: wayned@samba.org
ReportedBy:
2009 Nov 25
2
Restricting transfers between SIP phones
Hello,
We are in the process of splitting our phone system into two separate
logical systems for our two departments. One of the goals of this
switch is to restrict members of one department from transferring calls
to the other, but not restrict them from calling that department
themselves. So what I need to know is how to detect whether a call
from a member of that department is a transfer or
2012 Jan 26
2
Too many open files
Hi all,
While trying to track down a T.38 issue, I came across a series of log
entries like this:
============================================================================
[Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr:
Unable to allocate socket: Too many open files
[Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create
socket
2008 Mar 17
2
Order of queue member list
We just recently upgraded from Asterisk 1.2 to 1.4, and quickly noticed
a change in the behaviour of the queues--a change that we cannot live with.
We've used AddQueueMember/RemoveQueueMember to manage logging into and
out of our queues for over a year now with Asterisk 1.2, and in that
version the queue members were sorted in such a way that the person who
had been logged in the longest
2015 Apr 29
2
PJSIP - sessions-timers support not working on 13.X
...um, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 28 Apr 2015 11:54:11 -0700
> From: Chad Wallace <cwallace at lodgingcompany.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] adding area code
> Message-ID: <20150428115411.71697421 at ws78.int.tlc>
> Content-Type: text/plain; charset=US-ASCII
>
> On Tue, 28 Apr 2015 07:21:12 -0700
> Motty Cruz &l...
2008 Sep 09
2
SIP to IAX?
Hi all!
I am looking for some software that would work as a proxy between a SIP
device (SIP phones and ATA boxes) and the Asterisk system running IAX. The
reason is that I can only get IAX trow the firewalls, and can't change the
settings.
One solution I am using are getting several Asterisk system communicate trow
IAX bout in this case would I rater have every persons computer act as a
proxy
2015 Apr 29
0
PJSIP - sessions-timers support not working on 13.X
...;> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>>
>>
>> ------------------------------
>>
>> Message: 3
>> Date: Tue, 28 Apr 2015 11:54:11 -0700
>> From: Chad Wallace <cwallace at lodgingcompany.com>
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] adding area code
>> Message-ID: <20150428115411.71697421 at ws78.int.tlc>
>> Content-Type: text/plain; charset=US-ASCII
>>
>> On Tue, 28 Apr 2015 07:21:12...
2010 Nov 14
8
dial plan and sip
Here is a very very basic config. But, not working (:
I simply want to dial the DID that is registered with the SIP provider.
then, as you can see the call should dial the 703111 number
Hints please?
sip.conf
;register => 908366554:396444 at carrier.jazzey.com
register => 908366554:396444 at sip.jazzey.com
[jazzey]
type=friend
host=sip.jazzey.com
username=908366554
secret=396444
2014 Nov 27
2
Strange Issue: asterisk deleted
...------
> An HTML attachment was scrubbed...
> URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141126/9deca244/attachment-0001.html>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 26 Nov 2014 14:54:27 -0800
> From: Chad Wallace <cwallace at lodgingcompany.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Strange Issue: asterisk deleted
> Message-ID: <20141126145427.4819c67b at ws78.int.tlc>
> Content-Type: text/plain; charset=US-ASCII
>
> On Wed, 26 Nov 2014 22:08:05 +0200
>...
2015 Apr 27
0
adding area code
On Mon, 27 Apr 2015 14:30:07 -0700 (PDT)
Steve Edwards <asterisk.org at sedwards.com> wrote:
> On Mon, 27 Apr 2015, Bryant Zimmerman wrote:
>
> > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1})
>
> Missing a colon?
>
> ${EXTEN:-1}
>
Does that work? I've always seen it like this:
${EXTEN:1}
--
C. Chad Wallace, B.Sc.
The Lodging
2015 Apr 28
1
adding area code
On Tue, 28 Apr 2015 07:21:12 -0700
Motty Cruz <motty.cruz at gmail.com> wrote:
> here is what I did and worked for me:
>
> exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444)
>
> exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80)
I find it hard to believe this is working.
First, you don't have a leading underscore on your patterns. Your
users
2015 Apr 09
0
dial out with channel variable; sub-string usage
On Wed, 08 Apr 2015 16:10:30 -0700
thufir <hawat.thufir at gmail.com> wrote:
> I want to do something like:
>
>
> exten => _NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN})
> exten => _Nxxxxxx,1,Dial(${BABY}/${EXTEN})
> exten => _1NXXNxxxxxx,1,Dial(${BABY}/${EXTEN})
> exten => _011.,1,Dial(Dial({TOLL}/${EXTEN})
> exten => _9NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN})