Re: [asterisk-users] Learn some terminalogy before mountingthistask.

[Ryan Amos]

The 7914 only works under SCCP; the SIP firmware does not support it at
all (the expansion panel won't even power on fully.) The SCCP channel
driver under Asterisk doesn't really support the 7914 very well,
currently it will only show onhook/offhook state (though there has been
much discussion recently about changing this.) If you want to do this
with SIP then you're better off with something like the grandstream
mentioned, or just use the Flash Operator Panel (IMO it gives you more
flexibility at a much lower cost.)
 
I have personally found "receptionist phone" functionality handled much
better with FOP. I have a 7914 and its functionality (and usefulness) is
very limited under Asterisk.



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James R.
Stevens
Sent: Monday, August 06, 2007 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Learn some terminalogy before
mountingthistask.



Thank you for your reply as it is exactly what we would need. Sorry I
didn't find it myself. I do have a question about configuration within
Asterisk. 

 

I'm reading the PDF on the Cisco Expansion module and it says 'When used
as a DN key buttons are illuminated ...'

 

Is that what we are doing within Asterisk or Trixbox when we configure
an extension?  (A Directory Number??)

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
FitzGibbon
Sent: Monday, August 06, 2007 7:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Learn some terminalogy before
mountingthistask.

 

On 8/5/07, James R. Stevens <[EMAIL PROTECTED]> wrote:

In the design of an Asterisk system using Cisco 7900 series SIP
phones
we are struggling with giving the reception folks (3) hardware
that can
tell them the status of everyone in the office (10 or so) (On
the phone, 
out of office etc) Something that would register each of the
extensions
we choose and give status of that ext.

What hardware (Phone or other) could we give the receptionist to
do
this?


You're probably looking for something like this:

http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0
9186a008008883d.html 

I have no experience integrating this specific piece of hardware with
Asterisk, but I've done what you're trying to do with the Grandstream
equivalent for our front reception:

http://www.grandstream.com/gxp2000.html

and

http://www.grandstream.com/gxp2000ext.html

As I understand it, so long as the device can do a SIP SUBSCRIBE for
each extension you want to monitor and you configure hints in your
Asterisk dialplan for those extensions, it should work.  You may need to
set 'subscribecontext' (in sip.conf) for the phone that will be watching
the extensions unless your hints are in the same context as the phone
uses for outbound dialing.

Of course, what the device does with the various payloads contained in
the SIP NOTIFY messages is going to be different for each phone.  On the
Grandstream I can see 'not in use' (solid green) and 'ringing' (flashing
red) distinctly, but 'unavailable' and 'in use' are both mapped to a
solid red, which makes it somewhat useless for transiently connected
user agents like softphones. 


Hopefully someone with experience will speak up and confirm that the
7900 series does interop properly with Asterisk for SUBSCRIBE and
NOTIFY.

If that doesn't work, you could always go with a software solution, like
the Flash Operator Panel.  voip-info has a list (look at the "Operator"
section on the page): http://www.voip-info.org/wiki/view/Asterisk+GUI

-- 
j. 


-- 
This message has been scanned for viruses and 
dangerous content by Athens Hyperion Scanner
 , and is 
believed to be clean. 
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RE: [asterisk-users] Asterisk on virtual machine

[Ryan Amos]

Asterisk does not work very well in a VM
due to the timeslicing. Dropped calls, jittery audio and echo can all creep in.

 

Good news is that an AD controller runs
just fine in VMware. Just make sure the box has enough RAM to keep it happy,
and use a physical second disk for the Windows install. So I’d suggest
running Asterisk in Linux as the native OS, and running VMware with Windows Server
as a guest OS. This setup should work just fine for you.

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
Sent: Tuesday, October 31, 2006
9:53 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] Asterisk
on virtual machine



 

We have a centralized infrastructure where
we deploy Asterisk servers in remote call centers for authentication and
transcoding. SIP g729a calls are then sent over an MPLS VPN to a central
Asterisk farm, from which calls are sent/received via PRI.

 

To avoid placing two servers in each call
center, one for Asterisk and another for Windows AD services, we have been
playing with VMWare. Can anyone provide their experiences in using Asterisk in
a VMWare configuration?  Good/bad/ugly?

 

Thanks,

Adam






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RE: [asterisk-users] Asterisk with cisco 7935

[Ryan Amos]

I spent quite a bit of time debugging the
7935/7936, and it is an issue inside the firmware that Cisco knows how to work
around in CallManager. There are better conference phone options available, and
development on chan_sccp is basically dead at this point anyway, so I don’t
see this one ever being fixed.

 

I would recommend a Polycom IP4000, it’s
the exact same phone body but is much cheaper MSRP, and it’s SIP.

 









From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Lacy Moore -
Aspendora
Sent: Tuesday, September 26, 2006
10:30 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users]
Asterisk with cisco 7935



 





Just wondering if anyone has had any luck getting the cisco 7935
working
with asterisk and if so, what is the best way to go about it?  on the






 





The consensus on the chan_sccp list is that it seems to be a good door
stop.  Seems something is just different about its SCCP image.  There
is new SCCP firmware that was released  this month.  I don't
know if it works any better. 




 






-- 
Lacy Moore
I'm the guy that doesn't give a damn about anyone's problems but my own... 






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RE: [asterisk-users] asterisk and PowerEdge 1950

[Ryan Amos]

It is almost always better to use a single T1/E1 card when possible to
avoid conflicts. A Digium TE2XXP series card sounds like what you would
need. The price is usually less than buying 2 single cards.

The server itself is fine. It has 2 PCI slots, so if you went with a
single card you would be able to expand later should you find the need.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stas
Khromoy
Sent: Thursday, September 21, 2006 1:22 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk and PowerEdge 1950

hey folks

we're planing to install asterisk for a client of ours
was just wondering if the Dell's PowerEdge 1950
will take 2 - T1 cards.

or if there any recommendations as to which 
server would be good for our project.


thanks

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RE: [asterisk-users] Cisco SIP Firmware

[Ryan Amos]

The 8.2 firmware works just fine (asterisk 1.2.6). I have had exactly
zero problems with it, nothing even weird about it. Pretty trouble-free
IMO.

I believe the phone that doesn't work quite right with the 8.2 SIP image
is the 7970. I have probably 20 7940s/7960s, all running the 8.2 SIP
image without problem. The server name in the caller ID is a little
annoying, but it is really just cosmetic (redial works just fine with
some dialplan stuff you have to do anyway.)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francisco
Gonzalez Canales
Sent: Thursday, July 06, 2006 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco SIP Firmware

Where may I find the 7.4 firmware for 7940? I was only able to find the
8.2
at cisco's website.

F


On 7/6/06 11:05 AM, "Aaron Daniel" <[EMAIL PROTECTED]> wrote:

> On Thu, 2006-07-06 at 10:55 -0500, Peder @ NetworkOblivion wrote:
>> What is the current recommended version of firmware for SIP on
>> 7960/7940's.  I was reading through some of the stuff on voip-info
and
>> it looks like the 8.x's have pretty serious bugs in regards ti *.
Thanks.
>> 
>> PA
> 
> We stick with the 7.4 firmware.  It does exactly what we need, doesn't
> decide it doesn't want to forget about registration if the server
falls
> out from under it, and doesn't have the server name in the caller id.


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RE: [Asterisk-Users] Cisco 7905G SIP firmware needed

[Ryan Amos]

To get the SIP firmware for these phones, you need to buy a Cisco
SmartNet support contract (about $75 USD in the USA, though I've heard
rumors a Europe-only contract exists for about $10 USD.) You can
purchase one through most Cisco resellers. That will give you access to
Cisco's download site. Configuration of these phones under SIP is not
quite as straight forward as it is with SCCP, but it's manageable.

SCCP does not work very well with asterisk in a large deployment setup
in my experience. I had approximately 25 phones on chan_sccp and the
stability was nowhere near where a commercial phone system should be
(we're talking 2 or 3 crashes a week) and is still lacking features to
make it useful to run an entire setup on (3 way calling, unattended
transfer, etc)

Unfortunately, I cannot get XML services working for the life of me
under the SIP image using the 7912G phones (which are essentially just a
7905 with a built in switch.) The configuration file schema has an
option for a services and directory URL, but the phone seems to ignore
them. XML services on the other Cisco phones like the 7940 or 7960 work
fine with the SIP image (but these phones use a totally different
provisioning method than the 7905/7912.)

Ultimately I'd say the recommendation comes down to how many phones
we're talking about and what kind of environment. Chan_sccp seems to
work fine in a small system where you're not going to be adding or
removing many users, but for any system you're going to have to support,
I would (and do) use SIP.

-Ryan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrea
Frigo
Sent: Thursday, June 29, 2006 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Cisco 7905G SIP firmware needed

Hi,
I bought a 7905G Cisco IP Phone and want to connect to Asterisk with
SIP 
protocol, but can't find a way to download this protocol update from
Cisco, 
Can anyone please help me?
Support the SIP protocol also the XML applications that I can use with
SCCP?
What is better, try to configure 7905G as SIP or try to use SCCP with 
Asterisk?

Best regards,
Andrea Frigo <[EMAIL PROTECTED]> 

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RE: [Asterisk-Users] 7960 help: transferring calls

[Ryan Amos]

Chan_sccp does not support blind transfer. I would suggest using
chan_sip and the SIP images with these phones; it is much more stable,
has more features and is being actively developed. Chan_sip supports
blind transfer and 3-way calling, plus it handles multiple calls on hold
a bit more gracefully than chan_sccp. Chan_sccp seems largely dead at
this point; the maintainer has not released a patch in 2 months and most
of the users who know the code well enough to possibly maintain it seem
to have moved on to other projects.

If you don't have access to the SIP firmware (which you can get for a
$75 cisco smartNET contract,) understand that chan_sccp still has quite
a few bugs that make it unsuitable for a production system in my eyes.
There are still unresolved deadlocks and channel locking issues which
can render your phone unusable until an asterisk restart, and you can't
reload the configuration without unloading the driver and killing
registration on all your phones (meaning you can't add a phone without
downtime for the whole system.)

But this is just my take on the situation.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Bagnall
Sent: Tuesday, June 27, 2006 9:57 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] 7960 help: transferring calls

Greetings all,

Not specifically an asterisk query, but a couple of transfer queries
that
I'm sure are obvious to folks who use these phones all the time:

1) how does one do a blind transfer? When a call is answered and one
hits
the transfer button, followed by an extension, one has to wait for the
other
party to answer, then hit transfer again, before the call is released.
I'm
sure there must be an option to answer a call, then fire it straight off
to
another extension without waiting for an answer?

2) if there are 2 incoming calls currently on the go (i.e. the first one
has
been put on hold for the operator to answer the second call), how does
one
determine which call will be transferred when the transfer button is
pressed?  Is there a way to select the source call for a transfer prior
to
hitting transfer?

3) when handling 2 calls, how does one swap between them?

These phones are running sccp through chan_sccp if that makes any
difference
to operation.

Thanks in advance folks.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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RE: [Asterisk-Users] Cisco 7936 Conference Phone - SIP or SCCP?

[Ryan Amos]

The 7936 is SCCP only. It kind of works
with chan_sccp, but it’s not something I would use in a production
environment (it’s very buggy and will stop accepting input from the keys
frequently, requiring a hard reboot.) I would say there is no good way to
reliably use the 7936 in Asterisk, and I wouldn’t consider it supported.
I’ve spent quite a lot of time on this issue, and even written a patch
that makes the phone sort-of-work, but the problem is that the phone doesn’t
like some of the data chan_sccp sends it and locks up. Killing the channel
while the phone is like this often results in a core dump.

 

If you need a SIP conference phone, check
out the Polycom SoundStation IP 4000. It’s basically the same phone as
the Cisco running a SIP image. I wouldn’t count on the 7936 ever really
being supported in asterisk, seeing as the Polycom is $300 cheaper and the
Cisco is just a rebranded Polycom with different software, plus you can get
support for the Polycom through a reseller for a lot less than with Cisco.

 

-Ryan

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, June 15, 2006
10:19 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco
7936 Conference Phone - SIP or SCCP?



 



Hi All,





 





Does anyone have any experience getting a 7936 to work with Asterisk?





 





Do you need to use SCCP or is there a SIP image for the phone?  I
have a few 7960G's and they are working with SIP, just curious if the config of
the conference phone is the same and if anybody has any good setup links. 





 





Thanks!





 





NB








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[Asterisk-Users] Zap DTMF detection

[Ryan Amos]

I am having some
troubles with DTMF detection on zap channels when the remote caller is calling
from a noisy cell phone. It is actually detecting multiple DTMF tones (usually
2 or 3) when only one is sent (i.e. I press ‘3’ and Asterisk is
detecting that as ‘333’.) I don’t know the exact situation
with regards to signal strength, etc, but it seems to happen more often from
users located in other cities. Here is my zapata.conf (I have tried with
relaxdtmf set to yes and no, with no difference.)

 

[channels]

usecallerid=yes

hidecallerid=no

callwaiting=yes

threewaycalling=yes

transfer=yes

echocancel=yes

echocancelwhenbridged=no

echotraining=yes

pickupgroup=1

immediate=no

callprogress=no

relaxdtmf=yes

 

signalling=pri_cpe

switchtype=national

context=default

rxgain=15.0

group=1

channel => 1-23

 

 

Has anyone else had
and solved this problem? 

 

-Ryan






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[Asterisk-Users] MeetMe, async password requirements...

[Ryan Amos]

Is there any way to
have MeetMe require a password normally, except when dialed through a certain
context or with a certain flag? As far as I can tell, there is no flag to
MeetMe() to disable PIN checks the way you can do with VoiceMailMain. This
would be a very useful feature, as through context usage you could require a
password from “external” users but not require them for “local”
users who call using a different context.

 

Does this feature
exist, or would I need to create my own access verification system around
MeetMe to get around this? I’d rather not have to do that. :)

 

 

-Ryan

 






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RE: [Asterisk-Users] *.conf utilities for Asterisk

[Ryan Amos]

I personally just use some custom perl scripts to parse a homemade
config file containing lines, MAC addresses of phones, group lines and
names for everyone (and several binary flags for specific features.) The
perl scripts parse the config files, merge them with my templates and
write the resulting files to disk. For any given phone, the only
information you *really* need is the MAC address and what lines it is
subscribed to.

It actually doesn't take too long if you're even remotely handy with
perl. I find that most pre-made asterisk configuration packages are too
limited for general use. There are a myriad of applications you can use
Asterisk for, and almost no 2 users will have the exact same setup. The
asterisk config files are actually really well thought out and very easy
to parse/rewrite with small perl scripts.

-Ryan
  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
mustardman29
Sent: Monday, May 08, 2006 3:12 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] *.conf utilities for Asterisk

 
Hi all,

I was wondering if anyone has any recommendations for *.conf generators
for
Asterisk.  Creating *.conf files manually for Asterisk requires too much
effort for what I do other than minor tweaking.  I run Asterisk as a
network
appliance (Astlinux on CF) so something like FreePBX that needs to run
on
the Asterisk server itself is not an option.

I have been using IPmanager up until now which worked great but
development
has been discontinued on that product.
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RE: [Asterisk-Users] Digium cards, so disappointing !

[Ryan Amos]

I am using a digium TE110P and a TDM04b (or whatever the one with 4 FXS
ports is called) on a Dell PowerEdge 2850. No problems at all with
faxing with a cheap fax machine, though the asterisk box almost never
goes above 5% CPU usage unless there are some conference calls going on.
I can run modems/faxes just fine (though the modem connections seem to
have a bit more latency than through a POTS line, it is acceptable for
our use.)

Just be sure to set echocancelwhenbridged=no and tweak your txgain and
rxgain on the line (this is not a do it once and you're done thing, I
had to go back probably 5 times over the course of 2 weeks to get the
right numbers.) I am even doing a redirect to eFax (I'd do with asterisk
but we already had an efax account and it works well enough) on one of
my DIDs and it works great.

Quite honestly I found a lot of documentation on how faxing in Asterisk
is hard, and I just never saw that. Maybe I got lucky with a magic
combination of hardware and forgiving fax machine, but it kind of just
worked the first time I tried it. Now, if only setting up a 'page all'
function on Cisco 79XX SIP phones without using a line appearance was so
easy...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Friday, April 14, 2006 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Digium cards, so disappointing !

Anton Krall wrote:
> Problem is, how to make sure you system WILL have 100% on zttest
before
> buying the cards.. You need to have stability, compatibility and
certainty
> that what you buy is going to work :(
> 
> Anybody had similar problems or success stories with sangoma cards? 

Running zttest on my box with both a TDM04b and A200D installed 
indicates and average of 99.96% for both. Not sure how accurate that 
might be as the A200D card appears as a 24 channel interface in terms of

/dev/zap even though only four ports are equipped.

The TDM04b won't support faxes on this box under any circumstances and 
I've played around with about every possible pci latency, etc, change 
that folks have suggested in the last two years.

Based on my heavily invested testing to date (which includes about two 
years of doing this), the "only" usable fax support thus far comes from 
using the A200D card with the fax machine directly connected to a fxs 
port on that card, and an fxo (pstn) port on the exact same card. Those 
fax tests have been 100% solid using a cheap/older Brother fax machine.



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RE: [Asterisk-Users] Cisco 7912 Phones & XML

[Ryan Amos]

This phone can be used
with the SCCP firmware and it does support XML services (you need a cisco
smartnet login to get the firmware.) However, the SCCP drivers available for
asterisk are not as mature (I have lots of random stability problems I can’t
track down,) and don’t have some key features like 3 way calling. 

 

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Mattina
Sent: Friday, April 07, 2006 10:17
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco
7912 Phones & XML



 

Friends,

http://www.voip-info.org/wiki/view/Cisco+7905%252F7912+IP+Phones
states under “SIP Software Limitations” that XML is not supported
on this phone.  However, my tech data summary of this product states: 

 

“In addition, XML applications deliver
impressive applications and network data to the Cisco IP Phone 7912G
display.”

 

Can someone please clear this up for me?

 

Thanks,

Adam Mattina
Networking & Systems Support
Layer 8 Group, Inc.
585.442.
[EMAIL PROTECTED]

 






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RE: [Asterisk-Users] chan_sccp and hinting

[Ryan Amos]

Seeing as chan_sccp is the only way to use the 7960+7914 sidecar
currently (chan_skinny is basically useless in a production
environment,) I believe it is relevant to this list. As far as I know,
you can't do ring notification with hints (though there may be another
way to do it with line presences; all I need the sidecar for is
transfers so onhook/offhook is all I want.)

Anyway, since when did this become the "asterisk-users as long as it is
distributed with asterisk" mailing list? I can understand not wanting it
on asterisk-dev, but asterisk-users?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Thursday, April 06, 2006 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] chan_sccp and hinting

> Next time you better ask chan_sccp related questions on the
> chan-sccp list

Shoot me for not wanting to subscribe to yet another mailing list when 
someone on here might have the answer.

Guess I won't ask if anyone's gotten ringing notification working on it.

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[Asterisk-Users] SIP caller id

[Ryan Amos]

I am using some Cisco
7940s with the 8.0 CM SIP image on them, and was wondering if there is a way to
have the caller ID display as just ”NAME” number as opposed to ”NAME”
[EMAIL PROTECTED].

 

The way it currently
is, the missed calls directory can’t be dialed, and my users really want
this feature. Is there any good way to strip the @ off of the caller ID? I
don’t see it being sent by asterisk is the reason… Any help is
appreciated!






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RE: [Asterisk-Users] Dell PowerEdge 2850

[Ryan Amos]

I use a PE2850 with CentOS 4.2 on it (as parent says, it is essentially
RHEL 4 without the support contract.) Extremely stable; no problems with
asterisk at all. Dell makes 2 PCI riser cards for this server, I believe
one of them has 5v slots. I have a 3.3v card so I can't tell you on
that.

-Ryan
  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: Monday, February 20, 2006 3:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dell PowerEdge 2850

Don't know about the Dell. I personaly use Cent OS
(www.centos.org) which is RHEL ES without paying for
it. I have it on my server and it seems to be holding
up just fine.


--- Richard OSS <[EMAIL PROTECTED]> wrote:

>   Hello,
>
>   Digium uses the Dell PE 2850 for their testing.
> This site says that 3.3V PCI slot.
>  
> http://www.voip-info.org/wiki/view/Asterisk+hardware
>
>   We are planning on purchasing a Dell PE 2850 and
> putting a TE205P card on it. However, the needs a 5V
> PCI slot. Does Dell PE 2850 has a 5V PCI slot? A
> person in our group tried to call Dell's customer
> support but they do not seem to know.
>
>   We will also be using RHEL ES 4 as the OS.
>
>   Anybody have experience (good/bad) for this type
> of configuration? We are going to use it primarily
> as a conferencing server serving 30-50 simultaneous
> users.
>
>   Can anybody recommend an alternative server that
> works well with TE205P and RHEL ES 4?
>
>   This is our first time using Asterisk so we would
> like to have it pain free as much as possible.
>
>   Thank you very much.
>
>   richard
>
>
> 
> 
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>   
>
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> 


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RE: [Asterisk-Users] Asterisk vs. Traditional PBX

[Ryan Amos]

TDM cards are for terminating regular POTS lines (anything you could
plug a regular phone into and have it work.) For a PRI you would need
one of Digium's T1 interface cards. The TE110P is a great card if you
need less than 23 channels. I can't recommend any other brands because
I've never used them.

_

RYAN AMOS
System Administrator

FINETOOTH
THE CONTRACT INTELLIGENCE COMPANY
phone  512.637.3530fax  512.637.3501 
mobile  512.484.6577
email  [EMAIL PROTECTED]
WWW.FINETOOTH.COM
  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nora
Lavelle
Sent: Friday, February 10, 2006 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk vs. Traditional PBX


Thanks everyone ! 

One more question if I do go with a PRI line. Will my existing TDM card
from digium work or do I need to purchase a different card to handle
this ? 

Thanks
Nora Lavelle


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid
Bender
Sent: Friday, February 10, 2006 4:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk vs. Traditional PBX

I am sorry. I thought you wrote Dell 850. Should have
looked closer. The machine should do just fine.
However it would not hurt to ptu in another gig. Also
see if anyone else on the list has used a 650 and what
expiriences they have had.

Regards,
Dovid
 
--- Nora Lavelle <[EMAIL PROTECTED]> wrote:

> 
> Hi Dovid, 
> 
> Thank you for the book. I'm already reading it. 
> 
> I have a dell 650 server, 1Gig of memory, 1 CPU
> (3.07Ghz).  What
> hardware would you recommend for the 200 users w/
> about 20 concurrent
> calls ? 
> 
> As always I thank you so much for your help. 
> 
> Nora Lavelle
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
> Behalf Of Dovid
> Bender
> Sent: Thursday, February 09, 2006 2:02 PM
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [Asterisk-Users] Asterisk vs.
> Traditional PBX
> 
> I think your problem is the Dell 650. What are the
> specs on it ? If you want a system that can support
> 200 users you will need to do a lot better than
> that.
> Also you will be dealing with T1's/E1's and not POTS
> lines. I think a good place to start (if you havent
> already) is the book that has come out a while back.
> I
> have it on my server at
> http://www.h6315.com/ast_book/
> 
> Regards,
> Dovid
> (I posted my server and not from the publisher
> becuase
> I do not know thier URL and I have email access only
> now.)
> 
> --- Nora Lavelle <[EMAIL PROTECTED]> wrote:
> 
> > 
> > Hi everyone ! 
> > 
> > So here's my question of the day !  I need to make
> a
> > decision on whether or not to go to a voip
> solution
> > or configure an existing pbx (norstar) that my
> > company has available.  We are a small startup.
> I'm
> > wanting a solution that will support up to about
> 200
> > people, with direct dial-in capability, up to
> about
> > 30 concurrent phone calls and good voice quality.
> > Right now I have an asterisk deployment with about
> > 15 people on it. We have sipura 841 phones. The
> > biggest issue currently is voice quality. lot of
> > complaints there.  I have a dell 650 poweredge
> > (single processory system), with a digium tdm400
> > card and 4 analog lines plugged into it. 
> > 
> > So here are my questions: 
> > 
> > * Is asterisk a good solution for my company ? or
> > should I just install the traditional pbx and look
> > to move to asterisk in a couple of years ? (I
> > personally would prefer asterisk cuz I'm a  unix
> > person not a phone person so from a manageability
> > perspective i would love this ) 
> > 
> > * If I were to go to an asterisk solution to
> support
> > about 200 people with the requirements above what
> > hardware platform would you recommend ?  I'm
> > guessing I'd need a PRI line and a different
> digium
> > card? Also would a 1cpu poweredge dell be enough ?
> > or would that have to be upgraded too ?  
> > 
> > If anyone is running an environment similar to
> this
> > that can provide help I would really appreciate
> > this. I'm having a hard time making this decision
> > and would love to hear anybody's experience in a
> > real time environment. 
> > 
> > Thanks again this list ROCKS! 
> > Nora Lavelle
> > 
> > 
> > > ___

RE: [Asterisk-Users] Dell PowerEdge 1800 and TE410P

[Ryan Amos]

Yeah, the reported Dell issues seem to be with the x600 series (2650,
1650, etc.) No issues at all on my PE2850s (other than having to talk
Dell into selling me a power cable so the FXS ports would work.)

-Ryan  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andres
Sent: Thursday, February 09, 2006 9:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dell PowerEdge 1800 and TE410P

Niall Hallett wrote:

>Yeah, I saw that compatibility list and the potential problem with the
>onboard ethernet controller. Did you disable it and use a pci based
card
>instead?
>
>Does anyone else run PowerEdge servers with the TE410P?
>
>Thanks,
>Niall
>
>  
>
We run the TE410 on the PowerEdge 1850 and it is rock solid.  We do not 
have problems with the onboard ethernet controller.

-- 
Andres
Technical Support
http://www.telesip.net


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RE: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?

[Ryan Amos]

This is turning into a sysadmin theory flamewar, but I think the main
point is that Fedora probably isn't the best thing to run on production
machines for QA reasons. This is because Fedora is more or less the QA
testbed for RHEL. CentOS is, for all intents and purposes (except a
little bug I discovered with large block devices >2 TB) the same as RHEL
without the support contract, so it is probably a better choice for a
server you want to keep working for a while.

Debian stable would probably work just as well (though IMO debian tends
to be a bit TOO old,) as would SUSE's stable release version. Just don't
use a "testing" release on a production machine. "yum update" (or
up2date, or apt) is pretty safe on "stable" release trees, but in the
testing releases you can run into problems with package dependencies,
versions, slowly updated mirrors... you get the point.

-Ryan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jens
Vagelpohl
Sent: Wednesday, February 08, 2006 4:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update
ornot?


On 8 Feb 2006, at 09:43, JP Carballo wrote:

> Alex Barnes wrote:
>
>> I think the "once it's working, leave it alone" advice is very sound
>> indeed :)
>>
>>
> A similar rule says "If it ain't broke, don't fix it."

Until you realize some script kiddie has exploited another Apache/ 
mod_ssl bug and is now remote-controlling your box.

There are no hard and fast recipes here. Neither the "automatically  
apply any and all updates" nor the "build and never look at it again"- 
policies should be applied without taking the specific situation into  
account.

If your box is on the internet you simply cannot forego updates.  
Period. If your box is completely walled off from the internet you  
can be lax about it (unless you have to worry about attacks from the  
inside).

The best policy is probably one that is halfway between the two.  
There are packages you only ever want to update "under parental  
supervision", like kernels. Then there are packages where you want to  
grab any update you can get ASAP, like Apache, or PHP, or SSH. Yum  
allows you to express this in its configuration, you can exclude  
packages from the automatic update.

I personally run a nightly script that uses yum to determine if there  
are updates. I apply them by hand. However, this is only feasible  
because it runs on just two machines.

jens

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[Asterisk-Users] TE110P Zaptel config questions

[Ryan Amos]

I have a TE110P that I
will be connecting to a T1 PRI. This seems pretty standard, but I am only using
7 channels for voice. It’s a shared voice/data T1; 7 channels voice, 16
channels data and 1 D-chan, it comes into a telco router and is split into a
voice PRI and an Ethernet connection. The 7 voice channels and one D chan are
the only things on the backside PRI. Does zaptel need any special configuration
for this sort of setup, or would the telco router handle the conversion and
restriction? Would I still define the bchan as 1-23 and dchan as 24? Or would
it be bchan=1-7 dchan=8?

 

This seems like a
pretty common product for most telco/ISPs to deliver to small businesses. I am
a system/network admin by trade, not a telecom engineer, so please excuse my
ignorance! I have not tried connecting it yet, as we need these phone lines
during the course of the business day, but I will be testing it tonight so I am
trying to ask any pertinent questions before I’m up to my neck in it. :)

 

-Ryan






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RE: [Asterisk-Users] Change Asterisk User

[Ryan Amos]

Use group permissions. Add the apache user to the asterisk group and
give the group the appropriate read and/or write access. IMO this is the
easiest way to get around the apache permissions thing, and probably the
Right Way (tm)

-Ryan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of amaury
BOSSE
Sent: Monday, November 07, 2005 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Change Asterisk User

Thanks for your answer,
I am working on Debian Sarge but I have compiled Astersik 1.0.9 myself
without .deb Packages.
I need to access to voicemail and sound files from my web-interface (php
and cgi/perl) but I can't change Apache user because of others
applications.
Asterisk creates files under Asterisk user and I have to access them
from www-data user.
Do you have other solution? I have tried using sudo but it doesn't seem
to work.

Regards,
Amaury


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RE: [Asterisk-Users] CentOS vs. Vanilla Kernel

[Ryan Amos]

The default CentOS kernel has worked fine for me.

Just an FYI; CentOS uses the RedHat EL kernel source to build... It's
pretty heavily patched so if you want to use the latest stable, download
the SRPMs from RedHat/CentOS and patch in the kernel.org patches.

But yeah, stick with the CentOS kernel unless you have problems. 

-Ryan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: Monday, November 07, 2005 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] CentOS vs. Vanilla Kernel

HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned
off).
OS: CentOS 4.2
Dual Embedded NIC enabled
USB disabled
serial disabled
printer disabled
2x73GB SCSI in HW Raid 1

What is the opinion of this fine list  - should I use the default CentOS

kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable 
(2.6.14)

Anyone got any clues / hints / tips on what should go into the kernel ?

All views and comments appreciated :)

Julian.

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[Asterisk-Users] Cisco phone firmware

[Ryan Amos]

I understand that I
must pay for a support license to download Cisco firmware, so I’m not
trying to pirate it. I simply want to know what I need to buy in order to get
firmware files for my phones. Does anyone have any helpful links they can give?
What does this license cost?

 

Specifially, I need
SCCP images for the 7902 and 7940… Does anyone have any experience with
setting this up and can either give me a hand or point me in the appropriate
direction? Most of the entries on voip-info.org were pretty light on details,
just postings of different config files. Also, if I’m just doing skinny
(no need for SIP, and the 7902s don’t support it anyway) do I even need
firmware files? It seems like I do, but I can’t even seem to get my
phones to try to tftp in and download anything.

 

Also, for anyone who
has used either of these phones, how well do they work with asterisk? The chan_sccp2
drivers say they “mostly work” but I want to know what doesn’t
work to see if I care. Any help would be appreciated, thanks.

 

--

Ryan Amos

System Administrator,
FineTooth

http://www.finetooth.com/


512-637-3530

 






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RE: [Asterisk-Users] Re: Anyone aware of a current Dell servermodelwith 3PCI slots

[Ryan Amos]

As a side note:

Don't expect to use a TDM400P card with any of the S110M modules; the
PE2850 does not have any internal power connectors to connect to the PCI
card. Kinda bummed me out.

And yes, you can (kind of) set the PCI IRQs in the BIOS. And Dell's
network cards do seem to have IRQ conflicts, though I was under the
impression that the 2850s were resolved with respect to this (the 2650s
do not work with Digium cards, but the 2850 should be fine.)

I haven't actually gotten asterisk working on this setup, mind you, I'm
still waiting on some VoIP phones to come in via UPS...

--

RYAN AMOS
System Administrator

FINETOOTH
THE CONTRACT INTELLIGENCE COMPANY
phone  512.637.3530fax  512.637.3501 
mobile  512.484.6577
email  [EMAIL PROTECTED]
WWW.FINETOOTH.COM
  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, November 02, 2005 4:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Anyone aware of a current Dell
servermodelwith 3PCI slots

I know of an install using a 2850 where it ran much better after they
disabled the onboard network card.

PaulH

- Original Message - 
From: "Tom Hayden" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, November 03, 2005 7:45 AM
Subject: Re: [Asterisk-Users] Re: Anyone aware of a current Dell server
modelwith 3PCI slots


> Yeah, with our Dell Poweredge 750, we had all kinds of IRQ conflicts
> and whatnot. I booted up and in the BIOS I turned off all sorts of
> devices, including one of the ethernet cards, the USB, serial, etc.
> After that, things worked much better.
>
> --
> Tom
>
> On 11/2/05, Matt <[EMAIL PROTECTED]> wrote:
> > >
> > > >We have a poweredge 2850 that we use for our VoIP server and it
has 3
PCI slots.
> >
> > > I'm greeting to hear this. I have installed some Digium cards into
this
> > > kind of servers.
> > > I get surprised when the slots pci gets shared IRQ with ethernet
> > > devices, raid controller or VGA card.
> > > Anybody knows how get unshare the IRQ of the slots pci ?
(firmware,
> > > update, some special BIOS configuration,...)
> > > We answered Dell with no response.
> >
> > I can't say that I've had this problem with the 2850 we have.   I
also
> > can't take the server down to look at it right now, however we just
> > got another digium card which I need to put in at some point over
the
> > next few days, so I'll be taking it down sometimes soon.
> >
> > As far as sharing, make sure you have disabled everything you don't
> > need USB, SERIAL, PARALLEL, etc.
> >
> > You can then set the PCI IRQ in the BIOS, I believe.
> >
> >CPU0   CPU1
> >   0:  584222710  584230408IO-APIC-edge  timer
> >   1:  0  7IO-APIC-edge  keyboard
> >   2:  0  0  XT-PIC  cascade
> >   8:  0  1IO-APIC-edge  rtc
> >  14:  0  2IO-APIC-edge  ide0
> >  38:67120198751310   IO-APIC-level  megaraid
> >  48:  318573642 37   IO-APIC-level  eth0
> >  77: 1014625786 2080170691   IO-APIC-level  t1xxp
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>
>
> --
> Tom
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>

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