Displaying 20 results from an estimated 100 matches similar to: "No subject"
2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
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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
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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
2011 Jan 10
0
No subject
takes precedence over a queue's defined moh class.
--=20
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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<div class=3D"gmail_quote">On Tue, Feb 1, 2011 at 10:20 AM, Danny Nicholas =
<span dir=3D"ltr"><<a
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 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 Nov 30
1
Polycom 500 format file system on every reboot
I have one client that is telling me that their Polycom 500's format the
file system every time they reboot, and also that they are unable to make
changes locally on the phone itself, only via the config files. If the
config file is not available when they try to boot the phone, then they
receive an error about not being able to find the config file and then the
phone will not boot up. Has
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 Feb 16
1
How does holdtime get calculated for queues
I had a customer ask me this question today, and I was surprised to say I
didn't have an exact answer for them. They have a relatively small support
queue for their business (three agents, and rarely more than one person in
line at any given time in the queue, if all agents are on a call), Their
support calls can range anywhere from 30 seconds to 30 minutes.
Occasionally, they'll get
2010 Jul 26
1
VPMADT032 Failed! Unable to ping the DSP (2)!
Running Asterisk 1.6.2.9, DAHDI 2.3.0.1, CentOS 5.5 (update to date as of a
week ago), I've installed a Digium AEX800P with 2 X400M FXO Modules and 1
VPMADT032 Module, hooked up to 5 analog lines. I get the error message
referenced in the subject in my dmesg output everytime I load / reload DAHDI
using the command "system dahdi start/restart". When I make an outbound
call over
2011 May 17
0
3. Re: ITSP Multi IPs (Alex Balashov) Asterisk-users Digest, Vol 82, Issue 33
Alex,
Thank you so much for your response. I've been so consumed with other
business that I only just now getting back to this issue. We have
implemented your suggestion which is perfect. Thank you again.
I've never asked a question of the community before and I'm extremely happy
with the rapid response I received.
Somewhat related to this initial problem I have an additional
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
2013 Jun 28
0
No subject
<br>
[memberconnector]<br>
;<br>
exten =3D> _XXX,1,Dial(SIP/${peerPrefix}$<u></u>{EXTEN},${TIMERINGQUEUE}=
,)<br>
=A0 =A0 =A0 same =3D> n,NoOp(DIALSTATUS=3D${<u></u>DIALSTATUS})<br>
<br>
As you can see, all status are empty,<div class=3D""><div><br>
<br>
-- <br>
2011 Apr 12
0
No subject
r>
<h2>Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010=
)
</h2>With SIP 3.2.X firmware (available on the Polycom download site)=20
and Asterisk 1.6.1, Polycom phones now support a full featured BLF=20
showing statuses of Ringing, Inuse and Online and one touch directed=20
call pickup.
<br>On the asterisk side all that needs to be done is to add a hint
2003 Aug 20
2
PRI CallerID problem
Greetings all..
We have an inbound/outbound PRI installed and terminated on a T400P ?
Digium Quad T1 card. We?re seeing an odd problem when sending
$CALLERIDNUM when calls from the PRI are forwarded back out to the PSTN
over the PRI. The $CALLERIDNUM is not being sent out along with the
call. It?s sending the phone number of the PRI itself, rather than the
$CALLERIDNUM information.
Yes, we can
2004 Aug 11
2
StanaPhone and Asterisks
I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them.
Here is what I have in my sip.conf:
[stanaphone]
type=friend
secret=pAsSwOrD ; skewed for this message.
username=3475341914
host=sip.stanaphone.com
2011 Sep 02
0
No subject
crashing.
So, as a first step to solving **that** problem, make sure asterisk is
compiled with debug
flags, dumps another core file, and then you do the "gdb asterisk
<corefilename>", and
get a stack trace. That should give us some idea of what happened.
>
> I have a fairly simple Followme sequence in place to see how it works
> before I get into the complex scenarios.
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
2005 Jun 21
0
Looking for PRI Outbound Caller ID Configura tion
As an employee in the technical operations of a CLEC this information is easily obtainable by anyone that has access to the Class 5 switch servicing that PRI... A Q.931 trace in the Class 5 Switch will tell the whole story....
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich Adamson
Sent: Tuesday, June
2011 Jan 10
0
No subject
Wait a second...
Action: DBGet\r\nFamily: DS\r\nKey: 0733025975\r\n\r\n
In the dialplan:
exten =3D> 0106024975,1,Set(DB(DS/0733025975)=3DINUSE)
exten =3D> 0106024975,n,Hangup()
exten =3D> 0106024976,1,Set(DB(DS/0733025975)=3DUNAVAILABLE)
exten =3D> 0106024976,n,Hangup()
Just a short call to my cell phone, to se if i get anything back, my =
cell phone doesn=E2=80=99t even ring.
Wait