Displaying 20 results from an estimated 1000 matches similar to: "No subject"
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 =
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2011 May 11
2
Asterisk SIP Trunking with Cisco UC 560
Hello,
I'm interested in knowing if anyone out there has successfully connected
Asterisk to a Cisco UC 560 via SIP trunking? We have a client of ours that
we put in an Asterisk install, one of their sister companies who we don't
control is putting in a Cisco UC 560. From my looking I think it can be
done, but the vendor is telling them it can't. Thought I'd ask around here
and see
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 =>
2009 Oct 06
2
Transfers from Queue Calls
Hello,
I thought to post this here before my manager starts his own coding
project to give us a workaround. My situation I'm running into is as
follows:
1. A call comes into our Asterisk system, it's trunked from one office
to another via IAX.
2. Call enters a queue and is picked up by one of the agents.
3. That agent has to transfer the call, could be for a number of
reasons the client
2009 Jun 18
2
Voicemail Password
Does anyone know of a way to force the voicemail password for users to be of a certain length? We've setup operator=yes within our voicemail.conf and want to have the users use a long password to prevent possible guessing by external parties. I'm not seeing any such option in my research. If it doesn't exist it might be a decent feature. Thanks.
Running: 1.4.25, on CentOS 4.7
2009 Mar 12
2
Timeout for Queue
Hello,
We had an incident recently where a call was in queue for an extended period of time. We use queuemetrics for reporting, and it reports that the call was waiting for 20 minutes. The different thing about it is that the disconnect reason is stated as Timeout. Is there a set maximum time a call will wait in the queue before being automatically disconnected? I tried looking through the code
2009 Jan 08
1
Goto Question
Hello,
I'm running three asterisk boxes, spread across three different countries. One of the offices is running Asterisk 1.2.18 on the Druid Telephony Platform(not my choice, has been in before I started and haven't had the time to remove it).
My situation I have is based on the contexts already in place, particularly for outbound calls, I need to do a Goto sending the call back into the
2009 May 26
2
Converting Cisco 7961 to SIP
As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files:
apps41.8-4-3-16.sbn
cnu41.8-4-3-16.sbn
cvm41sip.8-4-3-16.sbn
dsp41.8-4-3-16.sbn
jar41sip.8-4-3-16.sbn
2009 Mar 20
3
OpenSIPS on CentOS
Hello,
I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via
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
Moh show files
This will show you if your class is set up correctly.
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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
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
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
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.
>
>
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
2007 Jul 12
0
No subject
1. Is it normal to see :
# lsmod
Module Size Used by
dahdi_dummy 3236 0
Shouldn't it be used by asterisk or is this 0 value meaning something
specific ?
2. How can you check dahdi is running ?
Here, "ps aux | grep dahdi " replies "grep dahdi".
Cheers
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2011 May 13
0
[LLVMdev] [ptx] Propose a register class naming convention change
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Justin Holewinski wrote:
<blockquote
cite="mid:BANLkTi=Y9EFmWRu-9dQxydq8zTyF7tEbJw@mail.gmail.com"
2009 Jul 20
0
No subject
-uzzi
PS: If you're not seeing any connection information, be sure to double-check
the IP address is correct. Learned that lesson the hard way =\
On Sun, Jan 31, 2010 at 5:51 PM, Jim Rosenberg <jr at amanue.com> wrote:
> Let's say I have two Asterisk boxes, A and B. I am trying to get A to do
> SIP registration on B, so an extension for A can dial SIP phones covered by
>