the Cisco 3925.
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asterisk
, Def Mem)
(1) AS54-AC-RPS-PWR, AS5400 AC Redundant Power Supply
(1) AS54-DFC-8CT1, AS5400 OCTAL T1/PRI DFC Card
(2) AS54-DFC-108NP, AS5400 108 Voice/Universal Port Feature Card
Any thoughts would be appreciated. Thanks.
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On Wed, Oct 5, 2011 at 2:04 PM, Kyle Sexton k...@mocker.org wrote:
Does anyone know if there is a resource to see what changes were made
between different versions of Digium cards? For example, how
different is a TE410P revision C when compared to a TE410P 5th
generation card?
I know
.. but when did those occur?
In other words, is there a card revision changelog?
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# will be able to connect
agentXPerms 0660 0550 nobody asterisk
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Something like this should work:
exten = _011.,1,Answer
exten = _011.,n,Wait(1)
exten = _011.,n,Read(password,enter-password,5)
exten = _011.,n,GotoIf($[${password} = 12345]?5:9)
exten = _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
exten = _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
Just to help with troubleshooting you could try to reproduce the same problem
with a different set of SIP endpoints. Setup a soft phone as the destination
and see if the problem occurs there. That way you can eliminate the handset as
a potential problem.
On Sep 15, 2011, at 10:31 AM,
I have no solution, but my head hurts thinking about listening to separate
calls simultaneously in each ear.
On Sep 9, 2011, at 9:22 AM, fhirschberg wrote:
Hi list!
I'm using the latest Asterisk 1.8.6.0 cross compiled for an i.MX27 board
and it works really good.
But I need a feature and
using:
Lucent Max TNT
Dialogic IMG 1010
Cisco (Not sure which model would be best for this, the AS5400?)
Any real world experience/advice using something like this would be
appreciated, thanks.
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Just a note to anyone in the Kansas City area that I've relaunched the
KCAUG website/group at http://kcaug.org. Please drop by and join the
group. :)
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in the
queue and I have to reload chan_agent.so in order to get them available.
I'm running Asterisk 1.4.17, and the bug sounds a lot like
http://bugs.digium.com/view.php?id=9618 but that bug looks to be fixed in
1.4.17.
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) is
that they provide the CentOS Plus repository, so installing
PHP5/MySQL would be something like:
# yum --enablerepo=centosplus install php php-mysql
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. It
certainly isn't a replacement for fixing the root causes of whatever
that makes asterisk die, though.
Sajith.
Another vote for Monit, easy process/resource monitoring that just
works.
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.
Don't think this is possible w/ Asterisk. Here's a link to a Trixbox
thread about it (short and sweet).
http://www.trixbox.org/forums/trixbox-forums/trixbox-endpoints/polycom-park-soft-key
I'd love to be wrong though!
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Does anyone know who really makes this phone:
http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/
Large pictures are at the bottom:
http://www.hybsys.bg/img/ipph/IP5000_1.jpg
http://www.hybsys.bg/img/ipph/IP5000_2.jpg
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or input been
made with 'background'. You can set up a 'exten = i,1...' to prompt for
wrong keypresses - insult the user and so on. So this wont work if
someone just dials somthing wrong.
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the extension is, so the call dies. It
doesn't go to voicemail like I would like it to because that extension
never proceeds through my dialplan.
Looking for suggestions on getting around this so I can keep deploying
soft phones to agents in the field.
Thanks in advance!
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actually reach?
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but don't ever go to
voicemail (Just Dial(SIP/))
2. If a call comes in for that same agent to their DID, route to
stdexten macro
3. All calls routed w/ DUNDi so the system doesn't care about which server
they log into
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http://www.mocker.org
to the list. In my mind I would rather have too many tests than
not enough! :)
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time, but maybe put a
regexten=12345 for user jwolosuk?
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occurs after the page goes out. Also, make sure that
the pages are being disconnected (so you don't have lingering channels
being used) and that you aren't doing *any* transcoding on the page out.
'show channels'
'uptime'
:)
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and find which server
an extension lives on.
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John Novack [EMAIL PROTECTED] writes:
There is no guarantee of no risk.
Your mother lied to you when she said everything would be alright
Maybe we can convince Digium to have an indemnification program for
people who purchase the business edition! :)
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by anything coming to that DID? I don't think 800 numbers actually pass
that they are 800 #s.
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!
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www.iaxtel.com seems to be down, does anyone know if there is another way to
register new numbers with them?
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everyone to call each other, but that's not generally too fun.
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is matching on the 64XX and not searching out to see if there is a
*more* exact match than the pattern match. Is there any way to get around
this?
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On 6/15/07, Anthony Francis [EMAIL PROTECTED] wrote:
Kyle Sexton wrote:
I have two servers setup to do DUNDi lookups against each other. The
scenario is that on server A, I have a wildcard match for extensions
64XX that rings to a local extension on the server. On server B I
have a 6442
of those things, wouldn't it work well for a centralized
voicemail system instead of a solution like NFS or weird rsync scripts?
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there are probably
errors! :)
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component
32 Protocol Discriminator: Q.931 (8) len=9
33 Call Ref: len= 2 (reference 195/0xC3) (Originator)
34 Message type: DISCONNECT (69)
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On 6/8/07, Kyle Sexton [EMAIL PROTECTED] wrote:
Having a problem w/ not getting CID name from a PRI. CID Name appears in
the PRI debug, but even after a Wait(4) it still appears after the phone is
ringing. Here is the relevant info from my PRI debug output. Line 4 is a
NoOp showing me trying
, facing Quivera)
12056 W 135th St
Overland Park, KS 66221
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Overland Park, KS 66221
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Started playing with 1.4 and I'm curious what uses people have come up
with for the Jabber integration? So far I can think of presence based
call routing, but I'm sure there are other ideas. How are YOU using
the new Jabber features in 1.4? :)
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get a
totally natural sound out of was Aastra and Polycom. I'm finally happy
with the sound. I really had a hard time finding a provider that
supported smaller fish like me.
Dan
Kyle Sexton wrote:
Does anyone have any experience with any SIP or IAX providers that
support E911? I'd love to convert
Does anyone have any experience with any SIP or IAX providers that
support E911? I'd love to convert entirely to Asterisk at my house,
but the lack of emergency dialing has been a major hold-up for me.
Thanks in advance for any suggestions!
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(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
Have you made sure there isn't a firewall in the way that could be blocking
your audio? You might need to punch some holes through to allow your RTP
stream.
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I was wondering if anyone knows a good way to play back a message
to a queue member before the agent is connected to the caller. This
would allow an agent to know what queue the call was coming from by an
audible Queue name.
Thanks,Kyle
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For those that are in the Kansas City area I would like to announce the formation of the Kansas City Asterisk User Group. You can find more information about the group at http://www.kcaug.net/
. Currently there is a mailling list for the group, but if enough interest arises we will probably look
Depending on how much you can get one for, you may opt for a PRI and then get a channel bank to wire all the apartments with phones. If you are looking at straight VoIP you will have to be concerned about 911 service, etc..
Thanks,KyleOn 7/7/06, augustynr [EMAIL PROTECTED] wrote:
Dean,
I
I have to say that from the screenshots it looks very impressive. It's also interesting to hear that you created a proxy for the AMI, I'm sure other projects may be able to make use of that as well. I'm not going to install it on my production server (yet), but look forward to trying it at home!
This isn't really a fix for the people missing calls, but one solution I found was to limit the amount of time a call rings for to a cell phone. If it doesn't answer in X seconds, then dial again. This isn't a perfect solution, but helps some.
On 6/6/06, Colin Anderson [EMAIL PROTECTED] wrote:
We
Could you just set the variable in the part of the dialplan where they enter the queue and then reference it here?Thanks,KyleOn 5/26/06, Massimo Nuvoli
[EMAIL PROTECTED] wrote:This is the problem:
two QueuesAgent logged in as agentcallback and member of the two queues.When a call come in the
I'm having a problem with my dialplan. I'm trying to rewrite the CallerID Name variable so that when a call comes in, it shows what queue the call is going to:exten = 1234,n,Set(CALLERID(name)=Queue1)This works fine for most calls, but when I call from my cell phone my name appears and it doesn't
The Logitech 350 is a good sounding USB headset. The only problem that I've noticed is the mute button goes on a little too easy. :)Thanks,KyleOn 5/24/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I have a logitech USB headset and a labtec USB headset, and love both. The Logitech has better
Slight derail, does anyone have a good bluetooth headset that will work with a cell phone and a PC at the same time? The ones I have tried only work with one at a time.On 5/24/06,
fpeeters [EMAIL PROTECTED] wrote:
Bruno de Assumpção Loureiro wrote: Plantronics CS50, or someone bluetooth( it's a
I am having a problem with createlink not wanting to be disabled in my agents.conf file. No matter what when an agent picks up the phone, it appends the filename. Is there something other than 'createlink=no' that I should be adding to my
agents.conf to prevent this?Thanks,Kyle Sexton
let me know if you have questions about my setup.Thanks,Kyle Sexton
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macro and go to
voicemail, etc.. Please let me know if you have questions about my setup. Thanks, Kyle Sexton
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Have you tried putting a Hangup in your extensions.conf?On 4/13/06, Min Hwan Chang [EMAIL PROTECTED]
wrote:Details:Asterisk 1.0.9 Zaptel 1.0
Dell P3 1ghz with X100P Clone Location: IndiaThis is an interesting issue where when I open up ZTMonitor, it shows the RX as being on. It seems that Zaptel
Have you tried something like:exten = 2,1,SetCIDName(QUEUENAME: ${CALLERIDNAME})exten = 2,n,Queue(QUEUENAME)On 4/12/06,
Steve Feinstein [EMAIL PROTECTED] wrote:
Thanks!, I will definitely take a look at that.We were hoping not tohave to do AGI in the client, but if we have to, we have
I've also had horrible experiences with the Asterisk plugin. The second I enable it, no one can log into their IM client anymore.KyleOn 4/9/06, Kerry Garrison
[EMAIL PROTECTED] wrote:
I tried the latest version of Jive over the weekend and I have to say it isa giant pile of crap. I did this on
I'm not exactly sure what you are looking for, but I know you can use GotoIf to do conditionals based off criteria in the astdb.KyleOn 4/4/06, Dan Journo
[EMAIL PROTECTED] wrote:
Hi,
Please take a look at the following extensions.conf:-
exten = _11,1,NoCDR()exten =
Evar,I tried to duplicate this on my system but wasn't able to. Do you have the same password set for both voicemail boxes? Have you already logged into both of them individually (maybe it's a cookie thing?)
KyleOn 4/1/06, Ever Zalazar [EMAIL PROTECTED] wrote:
Hi everybody..I have the
Try sip show registry from the asterisk console.Kyle
On 4/2/06, Giridhar Reddy Bandi [EMAIL PROTECTED] wrote:
HII have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ?i generally look at the
Is the SIP phone behind NAT? That's one of the common reasons for one way audio. You might want to try forwarding some port ranges if you are behind NAT just to eliminate that as a possiblity. The SIP port ranges should be something like:
SIP: 5060-5061RTP: 1-2KyleOn 4/1/06, Il Neofita
Wow, no responses in favor of either the TE406P or the TE410P w/ a motherboard recommendation. That's not a great sign for this endeavor. :(Thanks,KyleOn 3/22/06,
George Pajari [EMAIL PROTECTED] wrote:
Kyle Sexton wrote: TE406P: - PRIs will go from up and working fine, to Provisioned, Down
So what motherboard/card combos do people use for high volume asterisk servers that are stable? What I'd need would be a 4-port PRI card, preferebly with echo cancellation, and a motherboard/system that works well with that card.
---Thanks,KyleOn 3/23/06, Kyle Sexton [EMAIL PROTECTED] wrote:
Wow
I am having a problem with asterisk not being stable enough for production use. I have two cards, the digium TE406P, and the TE410P. The TE410P is the primary card that I am using but I would like to move to the TE406P for the echo cancellation and more flexibility of PCI slots available.
General
I have the same setup and I don't have the problem you are having. The only difference I can see between my setup and yours is that instead of 'Playback', we are using 'Background'. Hope this helps.Thanks,Kyle
On 3/22/06, Sheeju .R.Alex [EMAIL PROTECTED] wrote:
Hi allI'm trying to dial out with a
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