Wonderfull ;)
On Mon, Apr 5, 2010 at 7:58 PM, Jason Parker jpar...@digium.com wrote:
bruce bruce wrote:
Thanks for the update Jason,
How do the upgrades work if v1.6.0 is already install and one wants to
upgrade to 1.6.2 (once it's available)?
yum upgrade asterisk*
???
Hello,
In asterisk-addons-1.6.1.2's changelog, you can see that 6 changes were
committed between versions 1.6.1.1 and 1.6.1.2.
But if I'm not mistaken, you cannot read anything there about Asterisk to
Asterisk-addons compatibility.
What is the rule for Asterisk to Asterisk-addons compatibility ?
On Mon, 5 Apr 2010, Warren Selby wrote:
On Mon, Apr 5, 2010 at 9:37 PM, Alec Davis siva...@paradise.net.nz wrote:
I've been asked for recommendations for a small call centre, an ethernet
SIP deskphone with a wireless headset.
Similar approach would be a mobile phone with bluetooth head
Several regulars from the VUC will be there, some of us are arriving
Tuesday night. Anyone else considering the trip? Post here or contact
me off list so we can meet.
/r
--
_
-- Bandwidth and Colocation Provided by
snip
Seconded - I've clients with the Plantronics CS60 and CS70's with the HL10's
on a combination of Snom and Grandstream GXP2000 desk phones. A bit extra
desk spaghetti, but they think it's worth it...
/snip
Seems like it's either 2 or 3 devices to make this work.
The lifter is not required, as
Randy R a écrit :
Several regulars from the VUC will be there, some of us are arriving
Tuesday night. Anyone else considering the trip? Post here or contact
me off list so we can meet.
/r
I'll be there but I don't know exactely when 'cause I'll at Paris this
week for my Microsoft course
I'll be there but I don't know exactely when 'cause I'll at Paris this
week for my Microsoft course
If you're on Twitter, follow @voipusers if you want to keep in touch
or email me if you prefer.
/r
--
_
-- Bandwidth and
Hi,
I have a problem with polarity reverse
this my dahdi config
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
James Lamanna wrote:
I'm seeing a lot of Exceptionally long voice queue length errors in
my logs, and then I seem to have a problem
where Asterisk will drop the registration for a significant number of
phones (they go UNREACHABLE), but then they
come back approximately a minute later.
Is
Hello list,
I need a hand to find the best dialplan failover solution when
using two SIP Trunks.
My reasons to do failover are:
a) one of the two providers could be unreachable
b) both providers may be UP but one of them could return a SIP error
I have been working on getting Asterisk and Exchange 2010 UM working together,
and so far I am pretty happy. The one thing not working right now is MWI.
I searched a bit and found this: https://issues.asterisk.org/view.php?id=13028
Now, please pardon me for being ignorant of all of this, but I
Is there a way to limit the number of simultaneous outbound SIP calls
made by Asterisk? We've tried using the 'Asterisk sip call-limit'
parameter but that doesn't seem to be working and one of our engineers
says that parameter has been depreciated.
Thanks,
deric.p...@nisc.coop
--
Here's one way - put your dial command into a macro that polls via a core
show channels and only dials when the count is below X. Even using a
slow language like PHP or PERL, an AGI call/return would not add as much
time to the dial process as PSTN delay does.
Example:
- exten =
Hi,
I do use the first solution based on DIALSTATUS variable. (
http://www.voip-info.org/wiki/view/Superdial+macro)
since it's included to a separated context named [superdial-macro], I don't
have to repeat it over and over, so the fact that it's not a oneliner
doesn't bother me at all :)
On
I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
the other fails:
-- Executing [...@macro-dialout-trunk:19] Dial(SIP/526-09eec7c8,
IAX2/InterOffice/210,300,tr) in new stack
-- Called InterOffice/210
-- Hungup 'IAX2/InterOffice-7578'
== Everyone is
Are there any way of configuring of Asterisk so it'll cache sound files
in memory, and when Asterisk receive a call, instead of loading sound
files from the disk
On Mon, 5 Apr 2010, Luki wrote:
Not directly, but it's not really needed. A long as the machine has
enough RAM, the files will
On Tue, 6 Apr 2010, Deric Page wrote:
Is there a way to limit the number of simultaneous outbound SIP calls
made by Asterisk? We've tried using the 'Asterisk sip call-limit'
parameter but that doesn't seem to be working and one of our engineers
says that parameter has been depreciated.
On Tue, 6 Apr 2010, bob gailer wrote:
I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
the other fails:
-- Executing [...@macro-dialout-trunk:19] Dial(SIP/526-09eec7c8,
IAX2/InterOffice/210,300,tr) in new stack
-- Called InterOffice/210
-- Hungup
On Tue, 06 Apr 2010 20:49:50 +1200, Alec Davis wrote:
snip
Seconded - I've clients with the Plantronics CS60 and CS70's with the HL10's
on a combination of Snom and Grandstream GXP2000 desk phones. A bit extra
desk spaghetti, but they think it's worth it...
/snip
Seems like it's either 2 or 3
I have a special requirement that insist an Asterisk server, 1.6.1.x,
is used.? I will have 2 SIP trunks coming into the server and I will
have to send calls to these SIP trunks with a round robin distribution
pattern.? I was thinking of using a group count function, if call
count is even
Thank you for your interest in my question and quick response. I am
relatively new to Asterisk, so I have a few specific questions regarding
your suggestions.
Then I will post to the list with a more meaningful subject and results.
On 4/6/2010 10:31 AM, Steve Edwards wrote:
On Tue, 6 Apr
Jay Vocaire wrote:
I have been working on getting Asterisk and Exchange 2010 UM working
together, and so far I am pretty happy. The one thing not working right now
is MWI.
I searched a bit and found this: https://issues.asterisk.org/view.php?id=13028
Now, please pardon me for being
On Tuesday 06 April 2010 07:46:05 Jay Vocaire wrote:
I have been working on getting Asterisk and Exchange 2010 UM working
together, and so far I am pretty happy. The one thing not working right
now is MWI.
I searched a bit and found this:
https://issues.asterisk.org/view.php?id=13028
Now,
On Tuesday 06 April 2010 03:16:45 Olivier wrote:
In asterisk-addons-1.6.1.2's changelog, you can see that 6 changes were
committed between versions 1.6.1.1 and 1.6.1.2.
But if I'm not mistaken, you cannot read anything there about Asterisk to
Asterisk-addons compatibility.
What is the rule
On Tue, Apr 6, 2010 at 12:36 AM, huu giang huugiang...@yahoo.com wrote:
Dear List,
Are there any way of configuring of Asterisk so it'll cache sound files in
memory, and when Asterisk receive a call, instead of loading sound files from
the disk, it will load from the memory and so Asterisk
I'm trying to build it and run into all sorts of problems. First,
make testexpr2 doesn't work at top level, nor in the main
subdirectory. If I manually try the commands for it in main/Makefile,
it doesn't have a main and calls ast_log. If use -DSTANDALONE2
instead, those go away, but then:
Did you tried the good old ram disk?
Flavio E. Goncalves
www.asteriskguide.com
-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de David Backeberg
Enviada em: Tuesday, April 06, 2010 12:50 PM
Para: Asterisk Users
Is the call successfull?
The 'Ignore polarity reversal on line seizure' may just be a warning.
What equipment, which Telco is the FXO card connected to?
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, April 06, 2010 9:25 AM
On Tue, 6 Apr 2010, Deric Page wrote:
Is there a way to limit the number of simultaneous outbound
On 4/6/2010 10:31 AM, Steve Edwards wrote:
On Tue, 6 Apr 2010, bob gailer wrote:
I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
the other fails:
-- Executing [...@macro-dialout-trunk:19] Dial(SIP/526-09eec7c8,
IAX2/InterOffice/210,300,tr) in new stack
call not succsessful.
I use nokia gsm gw witch have polarity reverse i try on my old asterisk 1.4.17
with digium tdm800 with fxo ports card polarity reverse works fine. But then i
connect to asterisk 1.6.2 with sangoma a400 with fxo ports card polarity don't
work.
polarity reverse is 600
On Tue, 6 Apr 2010, bob gailer wrote:
Verify the username, password, context, and extension all exist.
I do not understand or see username.
I do not see context.
I do not see or have passwords or know how to specify them.
Extension exists. I can call the other way with no problem.
Sorry
On Tuesday 06 April 2010 10:56:56 Richard Kenner wrote:
I'm trying to build it and run into all sorts of problems. First,
make testexpr2 doesn't work at top level, nor in the main
subdirectory. If I manually try the commands for it in main/Makefile,
it doesn't have a main and calls ast_log.
Why aren't you using check_expr in the utils directory?
Aren't they two different things? I thought check_expr looks at a whole
file for syntax errors while testexpr2 just parses one expression and
returns its value. But if testexpr2 doesn't exist anymore, shouldn't
the documentation be
Hello gang,
Is there a piece of software out there that can validate a
dialplan before I run it though my asterisk (1.4 and 1.6)? Right now I'm
just doing live run-time debugging, but that's slow and not always accurate
and my dialplan now exceeds 2000 lines. Any ideas?
Hi,
I'm running Asterisk 1.4.26.3 and I've noticed an interesting problem
when trying to play a Busy tone over a IAX trunk from the PSTN.
It seems as though Busy(20) returns non-zero immediately (it does not
wait 20s), so the caller never hears the busy tone, but
the call just appears to hang up.
Does TDM800 with FXO ports work with 1.6.2?
You should have also got other 'polarity related messages' during the call
setup.
One in particluar which prints debug info when a DAHDI_EVENT_POLARITY get
fired.
Code below.
ast_debug(1, Polarity Reversal event occured - DEBUG 2: channel %d, state
https://issues.asterisk.org/view.php?id=6643
CP
On Thu, Apr 1, 2010 at 7:36 AM, Ondrej Valousek webs...@s3group.cz wrote:
Hello,
Did anyone manage to force asterisk to put Remote-party-ID attribute on
the SIP outgoing call? I.e. When A calls B, I want that A gets a name of
B displayed on
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