use the qualify= syntax in your sip.conf and make sure it exceeds the
latency between the phones and asterisk server in ms.
-Greg
On Wed, 2005-09-28 at 16:17 -0700, Edwin Lam wrote:
hi folks.
we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and
we start having problems
Are you using CCM to operate your gateway with MGCP? If so, I had to
change the default timers under CCM advanced setup for Media exchange
timers or the call was timing out at 4 seconds. If the setup was
complete prior, it worked fine, but after 4 seconds q.931 from CCM would
tear down the
IMHO - you should not use price and quality in the same sentence for BV.
On Mon, 2005-09-26 at 15:20 -0400, Jason Schafer wrote:
I'm relatively new to the whole VOIP game, here's what I want to do. I
am using VOIPJet for all of the outbound calls on our AAH box. I have
one landline that I
Using 'dial-peer voice voip' and 'dial-peer voice pots' is
what you are looking for on the 7206..
Use sip, and just use dial(SIP/[EMAIL PROTECTED]) to make calls
go out..
search on cisco for dial-peer syntax and requirements - plenty of
examples out there.
-Greg
On Mon, 2005-08-08 at
Router#conf term
Router(config)#voice class codec 99
Router(config-class)#codec preference 1 g711ulaw
Router(config-class)#codec preference 2 g729br8
Router(config-class)#codec preference 3 g729r8
Router(config-class)#end
Router(config)#dial-peer voice 2000 voip
A hot thing in databases right now are clusters.
Has anyone setup a linux cluster and installed asterisk on it?
I assume you mean replication - and yes, we use that environment here
and it works well in most circumstances with HA running the failover..
We do not do much call processing, but
it as of yet, but it does not require
openh323 and pwlib.. That combo and gnugk on same box may work well??
It is relatively new though.
-Greg
On Wed, 2005-06-29 at 14:28 +0100, Barney Sowood wrote:
On Sat, Jun 25, 2005 at 07:58:24PM -0500, Greg Oliver wrote:
That works well. You may also want
, and the Hotel's connectivity is the last
word,
then my Vonage ATA should be choppy, as well, no? This is what leads me
to
think I can do some tweaking
later,
Paul
- Original Message -
From: Greg Oliver [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
Nothing you can do on this one.. Without the provider accepting your
QoS settings, you are at their mercy. And yes, you are correct, most
multi-tenant dwellings use xDSL for their connectivity due to it's
price, and the upstream is usually less bandwidth than the downstream..
-Greg
On Tue,
We have successfully connect * .9x 1.0.x with CCM 3.3.x and up using
both gatekeeper and no gatekeeper.. Using SIP usually with CCM 4.0 and
up.. With CCM 3.3.x, there is a limitation where the gateway H323 in
your case cannot use IP addresses, so the Asterisk box has to have
correct DNS entries
OS79XX is not used in 6 versions of firmware.. You can also, safely
go straight to 7.4.
Thanks,
Greg
On Fri, 2005-06-24 at 08:13 -0400, Tom Rymes wrote:
For what it is worth, this is not what I did. Since you have already
upgraded two phones to 7.4, I will assume that you know how to do
I have had several issues flashing between SCCP/SIP/MGCP on those phones
where it will eventually cause the handset to bleed through the
speakerphone. Once that happens, the phone is basically trash - it
never stops...
-Greg
I'm having a problem with one of our 7960. They all run latest
We use it like this if that is what you are looking for:
exten = s,4,GotoIfTime(8:30-17:00|mon-fri|*|*?open,s,1)
-Greg
On Wed, 2005-06-08 at 11:24 -0400, Henry Coleman wrote:
This feature is called attendant - night answer position. Is it not
possible to switch the incoming call to an
You can use super-valet-parking
On Tue, 2005-06-07 at 06:18 -0500, Mike Holloway wrote:
Reading through the code, I don't see a way of exiting the transfer and
regaining the call with the customer, unless the third party hangs up or
maybe doesn't answer and the dialplan doesn't do anything
You can try the ${RDNIS} variable.
On Tue, 2005-06-07 at 00:32 +0200, Elwin Andriol wrote:
Hello all,
This email is intended rather informative than questioning. While
developing some script-generated dial plan, we figured out that there
are differences between Snom 190's and BudgeTone
On Tue, 2005-05-17 at 09:26 +0100, tim panton wrote:
On 16 May 2005, at 22:54, Jean-Denis Girard wrote:
Andres Paglayan a écrit :
File::copy does copy, it re-writes the file,
you need to move it.
so when the the pointer is placed the file is already there.
Well from File::Copy
We use it for a client in 2 ways...
exten = _218X,1,SuperValetParking($[ ${EXTEN} + 100 ]|mylot|15|$ [${EXTEN} +
100]|10|superpark)
exten = _218X,10,Playback(vm-nobodyavail)
exten = _218X,11,Dial(SIP/${OPERATOR},15,m)
exten = _218X,12,Hangup
exten =
The only thing I have different in my CME dial-peers is application
session for each of them. Other than that, what you have works for me..
-Greg
Nathan Alberti wrote:
I have the following configuration and am obviously missing something
small that is causing * not to work as expected.
I have
True, but it also states that with no timeout value that it will dial
until the caller hangs up.
I have included my dial pattern - can anyone see anything that would
cause this, or something in my sip.conf or h323.conf files that would
override these settings?
Thanks,
Greg Oliver
[outbound
get no one is available to answer at
this time on the consoel and it redirects to an extension in
extensions.conf under a different context.
Any ideas on where I should be looking:
Thanks,
Greg Oliver
configs follow:
sip.conf
sip*CLI
sip*CLI
sip*CLI exit
Executing last minute cleanups
[EMAIL
. If I answer b4 the 5 seconds - everything is good.
Anywhere I need to set to get around this.
I have tried the t,T settings (even though the docs say no entry is
forever) with no luck.
Thanks,
Greg Oliver
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Asterisk-Users
OK - I have an cisco MGCP gateway controlled by a Cisco Call manager.
There is a gatekeeper controlled 323 trunk between callmanager and
asterisk (we use asterisk for vmail, sip phones, park/pick/meetme, etc..).
We are using GK routed mode in gnugk. All extensions in h323.conf and
callmanager
The 7.3 zip file contains the wrong filenames from their website. If
you watch the status messages, you should see a incorrect loads or
invalid loads flash before the phone continually reboots. I cannot
remember exactly which files I renamed to make it work, but I edited the
.loads file,
Turn on debug isdn q931 term mon on your 5350. It is an ISDN
signalling error. Strange it is showing up in asterisk through a 323
trunk though...
What happens when you do a csim start xxx where xx = phone
number to dial from the 5300?
-Greg
Tola Ogunsan wrote:
Hi Michael and
..
Any Ideas?
Thanks,
Greg Oliver
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I am using 7.3, and do not experience this behavior on Linksys or Cisco
powered switches not configured for voice vlan - I have a pre G model of
the phone though. I just cannot get mine to authenticate :(
-Greg
Mark Johnson wrote:
Asterisk wrote:
when booting the cisco 7960 with SIP image 7.3,
shell I get that t is registering, but not authenticated .1.
from show reg.
Any ideas would be appreciated. Only passwords were removed.
Thanks,
Greg Oliver
I have included my SIP.cnf file for review..
# SIP Configuration Generic File (start)
# Proxy Server
proxy1_address: sip.cistera.com
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