Displaying 20 results from an estimated 1000 matches similar to: "No subject"
2011 Apr 12
0
No subject
be able to setup a SIP trunk. I've been able to successfully integrate a
Cisco CallManager 7.x system with Asterisk using SIP trunking, so I imagine
you should be able to do the same here.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
--0016e651f0a6bbe47b04a303939e
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable
<div
2011 Apr 12
0
No subject
[0004f2xxxxxx](poly650)
defaultuser=0004f2xxxxxx
callerid="Front Desk" <1600>
mailbox=1600
*setvar=callidnum=1234561600*
and from extensions.conf:
[outgoing]
; Outbound unrestricted domestic calls
exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN}
on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.)
*exten =>
2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
------=_NextPart_000_016C_01CBF83B.306A1A90
Content-Type: text/html;
charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable
<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2013 Jan 16
1
Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start
I'm trying to decide if I need to open an issue for this or if it's just a
misconfiguration issue of some sort. Here's the situation - yesterday
morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS
5.8 installation and got a shell of a basic asterisk install setup (minimum
required configuration files, etc, with no dialplan or sip peers setup
yet). In the
2007 Jul 12
0
No subject
an external program, which at this stage, is not customizable ...
I don't know if alternatives (LiMO, Android, ...) would be more open to this
customization but for Symbian, not only Nokia SIP client wouldn't let you
autoanswer to SIP calls, but any other SIP client complying to Symbian
design wouldn't support autoanswer.
PS: Please, note that I'm far from being an expert in GSM
2011 Jan 06
0
No subject
If you don't use 'CERTVERIFY 1', then this will at least make sure that
nobody can sniff your sessions without a large effort (...)
> So, do I misunderstand CERTVERIFY directive ? Or is there a bug ?
>> Can you reproduce such behaviour ?
>>
>
> I'm not sure what is going on. Can you try running 'upsmon' with debugging
> enabled? The following are
2011 Jun 25
1
Cisco IP Phones and Skinny in asterisk 1.8.4.2 "tooooooooooooooooo"
Hi All;
Again, the Cisco IP Phones 7942G and using Skinny:
I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file.
The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination
2009 Jan 16
0
No subject
"Why Siphon doesn't allow to receive a call when it doesn't run
Apple doesn't accept (for the moment) an application runs in the background=
.
So, when Siphon doesn't run, the SIP server of your provider doesn't know
your iPhone."
--0015174c3c60a73ef5046656ca27
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable
2007 Jul 12
0
No subject
That's the main reason I opened this thread as it surprised me a bit ...
>
>
> Any 2-wire analog leg will be a source of echo. Many, many, many calls
> do not have a 2-wire leg.
Even in handset audio circuit ?
I was thinking that any handset is a potential echo source due to this audio
circuit ...
Do you agree ?
> Think cell/mobile or endpoints with PRI or T-1.
>
>
2007 Jul 12
0
No subject
such file or directory" on pure-IP platform in which I installed
asterisk-libpri-dahdi trilogy.
Maybe, it's me while following README instructions, maybe README
instructions could be improved or maybe it's wrongly labeled messages ?
That's why I told myself : I'm waiting too much from doc ? is a pure-IP
platform too specific ? what is the official policy ?
README starts with
2013 May 01
1
multiple provider for incoming
Matt,
At some point you need to consider how much is too much...
I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely
2009 Jan 16
0
No subject
...
Thanks, anyway for telling as at least, it reflects your needs.
>
>
> You want NT PtMP and i second that,
>
not being limited on the asterisk
> side is a must in the
> telephony ecosystem, since the legacy PABX aren't alwsys easy to
> reconfigure.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by
2011 Apr 12
0
No subject
supported, beside Idle, On call and Ringing ?
Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE |
INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD)
> Might be worth seeing if other phones do the same.
>
> S
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by
2009 Jan 16
0
No subject
connecting legacy PBX to Asterisk (for the very same reason, those PBX use
TE-PTMP).
If others could join this thread and say if they agree or not with NT-PTMP
being the 2nd most needed mode, would be great.
Please, do not hesitate to comment.
>
>
> Right now, I would not preclude the possibility that NT-PTMP support
> might be added, but I could not give you a concrete time at which
2009 Jul 20
0
No subject
mailboxes).
Are you certain that removing either 612 or 610 mailbox would keep Asterisk
from complaining ?
>
> However, the MWI does not indicate voice mails for 610 and I keep seeing
> this error message:
>
> ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox
> 610 in context a10
>
> However, mailbox 610 is clearly defined in voicemail.conf:
>
2011 Apr 11
3
changing port 5060 to 5061
please send me the ways to change asterisk port from 5060 to 5061 i
need to configure it because we are already using 5060 port in router
then we cant use it again we have to configure other sip server so
please suggest me a way..........................
On 4/10/11, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
> Send asterisk-users mailing
2011 May 13
0
[LLVMdev] [ptx] Propose a register class naming convention change
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=UTF-8" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
Justin Holewinski wrote:
<blockquote
cite="mid:BANLkTi=Y9EFmWRu-9dQxydq8zTyF7tEbJw@mail.gmail.com"
2009 Jan 16
0
No subject
could be "hot". Is there any chance this would cause the card to fail after
a while? It appears this site just had 4 port Digium card fail today.
> Also, I am trying to cross connect with another Asterisk system with
> > the normal LBO setting (i.e. span=1,1,0,esf,b8zs) but as of yet the
> > systems aren't seeing each other at all. Could the side with the high
>
2010 Jul 23
1
Attended Transfer question
I've been asked to implement the following transfer workflow in an asterisk
system, and I'm not seeing an easy way to do the bolded steps below (steps 4
and 5 for those with a text-only email client):
1 - Put the call on hold
2 - Call the extension for the staff member needed
3 - Give them a rundown of the caller and situation
*4 - Bring the caller on with the staff member the call will
2010 Jul 13
3
STRFTIME function declared in globals context
I'm trying to declare a few date-related global variables to ease my
dialplan. When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH = ${STRFTIME(${EPOCH},,%m)}
DAY = ${STRFTIME(${EPOCH},,%d)}
TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
If I evaluate these variables in the dialplan later, using
exten