Hi,
*yum install flite-devel* command is not giving any package in CentOS 5.6
32bit machine.
But the same command work on CentOS5.5 64bit machine.
Is any other package is required ?
On Fri, Apr 22, 2011 at 6:43 PM, Doug Lytle supp...@drdos.info wrote:
Satish Patel wrote:
from where I get
Does this ConfBridge requires a hardware timing source? Will I be able to use
this on any virtual server without having the need special changes to the VM
setup?
Thanks
C. Savinovich
On April 25, 2011 at 10:27 AM David Backeberg dbackeb...@gmail.com wrote:
On Mon, Apr 25, 2011 at 9:38
On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich
c.savinov...@itntelecom.com wrote:
Does this ConfBridge requires a hardware timing source?
No, and neither does MeetMe with modern DAHDI.
Will I be able to use this on any virtual server without having the need
special changes to
the VM
- Original Message -
From: David Backeberg dbackeb...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 25, 2011 9:27:19 AM
Subject: Re: [asterisk-users] The new ConfBridge application is now in
Asterisk Trunk!
- Original Message -
From: David Backeberg dbackeb...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 25, 2011 9:49:05 AM
Subject: Re: [asterisk-users] The new ConfBridge application is now in
Asterisk
On 11-04-25 10:49 AM, David Backeberg wrote:
On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich
c.savinov...@itntelecom.com wrote:
Does this ConfBridge requires a hardware timing source?
No, and neither does MeetMe with modern DAHDI.
This is the issue the OP was referencing. MeetMe depends
On Mon, Apr 25, 2011 at 11:10 AM, Paul Belanger pabelan...@digium.com wrote:
On 11-04-25 10:49 AM, David Backeberg wrote:
On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich
c.savinov...@itntelecom.com wrote:
Does this ConfBridge requires a hardware timing source?
No, and neither does MeetMe
Is the new conf bridge going to be in 1.8? or only 1.10?
Jerry
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- Original Message -
From: Jerry Geis ge...@pagestation.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 25, 2011 10:17:41 AM
Subject: [asterisk-users] new confbridge
Is the new conf bridge going to be in 1.8? or
As I see in your iax.conf, IAX Peer belogs to special context, which means
444 is allowed to make calls to extensions only on the same context
(Extension 111), can you call extension 111 ?
may be the other extensions are in the default context and you can
receive calls because extension 444 (dial
No, conference scheduling is not a feature that we have built
directly into ConfBridge, and I'm debating on what it would look
like.
Scheduling isn't built into MeetMe either, but the fact that it
dynamically reads from a database means that you can write external
programs (such as Web-Meetme)
I am trying to use the FILTER() function to strip out / from a CID
name. I have the following in my extensions.conf where I want to
perform the filtering:
exten = s,n,Set(NAME=${FILTER(\x20-\x2e\x30-\7d,${DIAL_NAME})})
However, when ${DIAL_NAME} is, say, J J DOE the string resulting
from the
On Mon, Apr 25, 2011 at 10:07:56AM -0500, David Vossel wrote:
On Monday, April 25, 2011 9:49:05 AM, David Backeberg wrote:
On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich
Does this ConfBridge requires a hardware timing source?
No, and neither does MeetMe with modern DAHDI.
Will I be able
Hi,
Please help me with this.
I am trying to setup a Class 4 termination setup using a kind of channel
hunting scenerio. I have some SIP DID numbers assigned from the local telecom
provider. My call comes from my wholesale client and lands on my switch, then
it is routed to asterisk. I want
If you want anonymous callers to be able to place calls to Asterisk, you need
to set allowguest=yes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul van der Vlis
Sent: Saturday, April 23, 2011 9:40 AM
Hi All-
I have successfully routed calls into our asterisk system from several DID
providers in the USA, but for some reason I'm having a problem getting Vitelity
to work.
We are using the IAX protocol, and the symptom is that only about 50% of the
calls terminate properly into my
Hi all.
When a user transfers a call by pressing the transfer soft button on their
phone, I'd like it to beep at them when the transfer is complete. I've got
it turned on in features.conf:
xfersound = beep ; to indicate an attended transfer is complete
xferfailsound = beeperr
Hi all,
Is it possible to send a SIP header to a PAP2T or SPA and cause the device
to automatically answer? I can do this with my Polycom phones and would like
to do it with my ATA's.
Any ideas?
--
Take care and have fun,
Mike Diehl.
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Monday, April 25, 2011 4:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Transfer beep w/ Polycom phone
Hi all.
On 4/24/11 1:21 PM, Bruce Ferrell wrote:
In the following example
exten = _1NXXNXX,1,Set(GROUP(outbound)=myprovider)
exten = _1NXXNXX,n,Set(COUNT=${GROUP_COUNT(myprovider@outbound)})
exten = _1NXXNXX,n,NoOp(There are ${COUNT} calls for myprovider)
exten = _1NXXNXX,n,GotoIf($[
i think i have similar problem after upgraded from 1.4.x to 1.6.2.17.
(originally upgraded to 1.8.3.2 unfortunately there were other more
pressing problems that forced me to downgraded it to 1.6.2.17)
i have a wanpipe device with 2 channels uses PRI signalling to PSTN
the other 2 uses FXO
On 23/04/11 8:45 AM, Tzafrir Cohen wrote:
So here's a mini poll:
Do you have a manager interface user that does not have all the read and
write permissions? If so: how have you managed to do so?
* Reading documentation / source
* An existing sample
* Trial and Error
Trial and Error - removed
No.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Monday, April 25, 2011 6:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PAP2T auto answer?
Hi all,
Is it possible to
You could try:
exten = *701,1,Set(__SIPADDHEADER=Call-Info:sip:192.168.101.1\;
answer-after=1)
--
Cheers,
Matt Riddell
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http://www.venturevoip.com/news.php (Daily Asterisk News)
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I had problems with a system I was trying to bring up using a couple older
a104d cards we had lying around. Neither card would pass audio. I worked with
one Sangoma tech for a couple hours while he tried various things. The second
tech I worked with got on the system and updated the firmware
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