Running in Xen domU does not in itself equal running in EC2. Yes, EC2 uses Xen,
but I'll bet your hypervisor's oversubscription/contention ratio is nothing
like theirs.
Amazon doesn't care about your PBX, it cares about squeezing every possible
dollar from the hardware nodes.
--
Alex Balasho
Hi Bruce,
We're running Asterisk in a domU Xen VM.
Works great, including conferences, but we can predict the availability
of hardware resources.
cheers,
Jan
On 15/07/11 10:57, Bruce Ferrell wrote:
> I'm relatively certain this is a silly question, but is anyone willing
> to share their experie
I run many things on Rackspace, but not my Asterisk servers. Reliability in
general (I don`t mean uptime, but the availability of CPU/Memory cycles for
your own needs) is either flaky, or much more expensive than buying your own
server would be.
Cloud-based should only be kept for things where la
We run on virtuals but not amazon
Sent from my iPhone
On Jul 14, 2011, at 6:57 PM, Bruce Ferrell wrote:
> I'm relatively certain this is a silly question, but is anyone willing to
> share their experience with asterisk in the amazon cloud?
>
> Does it work? Or do timing issues mess with audi
On 07/14/2011 06:57 PM, Bruce Ferrell wrote:
Does it work? Or do timing issues mess with audio?
It does, to a point. You will definitely see problems at higher
concurrent call volumes, and there is no question that concurrent call
limits reachable in such an environment are lower than on ba
I'm relatively certain this is a silly question, but is anyone willing
to share their experience with asterisk in the amazon cloud?
Does it work? Or do timing issues mess with audio?
Bruce
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On Thu, Jul 14, 2011 at 12:07:05PM -0700, J Gao wrote:
> On 11-07-14 11:42 AM, Barry Miller wrote:
> >On Thu, Jul 14, 2011 at 10:51:02AM -0700, J Gao wrote:
> >>Sorry for hijack this topic, but I have a different question:
> >>
> >>Every time I install Asterisk I have to "make menuselect" and to
>
The AstLinux Team would like to announce the immediate availability of the
0.7.9 release. This release includes either Asterisk 1.4.42 or
Asterisk 1.8.4.4. All current users are encouraged to upgrade to this release
to take advantage of bug fixes and other updates to Asterisk.
A full changelog
On 11-07-14 11:42 AM, Barry Miller wrote:
On Thu, Jul 14, 2011 at 10:51:02AM -0700, J Gao wrote:
Sorry for hijack this topic, but I have a different question:
Every time I install Asterisk I have to "make menuselect" and to
select/deselect some items. Now every time I have to write down what I
On Thu, Jul 14, 2011 at 10:51:02AM -0700, J Gao wrote:
> Sorry for hijack this topic, but I have a different question:
>
> Every time I install Asterisk I have to "make menuselect" and to
> select/deselect some items. Now every time I have to write down what I
> selected for future reference. I
Hi
First thank you for your help.
So, i need T.38 enable to use FAX; this site "
http://www.voip-info.org/wiki/view/Asterisk+T.38"; talk that works "Asterisk
*1.4* supports only T.38 fax pass through;", but, how enable it?
2011/7/14
> **
> Hi,
>
> T38udptl is T38 passthrough mode (T38-ATA -> As
Hi,
T38udptl is T38 passthrough mode (T38-ATA -> Asterisk -> T38
Gateway), Asterisk doesn't support T38-Gateway mode.
See
http://www.voip-info.org/wiki/view/Asterisk+T.38
and
http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway
Markus
On 11-07-14 09:54 AM, Robert Huddleston wrote:
I stand amused that people want to experiment with VoIP and Asterisk - but
aren't willing to:
( a ) Read wiki / manuals / faqs
( b ) demand packages for their o/s
This ain't windows folks :)
./configure
make
make install
Is really simple :)
-
Hey guys
I need of some help...
How i know if T.38 is enable on asterisk?
I saw the file /etc/asterisk/sip.conf and
/etc/asterisk/sip_general_custom.conf, both have the in "t38pt_udptl=yes";
Thanks!!
--
Eduardo Carpes
E-mail: car...@bsd.com.br
www.freebsd.org
--
__
I stand amused that people want to experiment with VoIP and Asterisk - but
aren't willing to:
( a ) Read wiki / manuals / faqs
( b ) demand packages for their o/s
This ain't windows folks :)
./configure
make
make install
Is really simple :)
-Original Message-
From: asterisk-users-boun..
On Thursday 14 Jul 2011, Kaushal Shriyan wrote:
> Hi,
>
> Any time line of availability of Asterisk binaries on CentOS version 6.
Yeah . as soon as someone compiles them :)
Really, building packages from source *IS* *NOT* *HARD*, and it doesn't even
take long anymore (on any target system
On 07/14/2011 09:40 AM, Kaushal Shriyan wrote:
Hi,
Any time line of availability of Asterisk binaries on CentOS version 6.
If you are referring to RPMs from packages.asterisk.org, they'll appear
in the near future. CentOS 6 just appeared a couple of days ago, so it
will take some time to get
- Original Message -
> Any time line of availability of Asterisk binaries on CentOS version
> 6.
The beautiful thing about open source software is that the instant Centos 6
became available, Asterisk was available to run on it.
Download [1], ./configure, make, make install, make samples.
Hi,
Any time line of availability of Asterisk binaries on CentOS version 6.
Regards
Kaushal
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Hello!
Is it possible to retrieve ISUP and other variables from the SS7
channel and use them in contexts in Asterisk?
Thanks,
Elliot
On Thu, Apr 15, 2010 at 6:04 PM, Ngo-Vi Hoai-Anh wrote:
> Sangoma uses wanpipe. Channel drivers set upon wanpipe (very
> simplifiedly speaking). You can configure
Hello
We have quite a strange phenomena in a genband c3 <=> asterisk 1.6.2.15 SIP
interconnection. (also reproduced with an 1.8 asterisk)
The Sequence:
c3 => Asterisk
Let's assume X < Y
We get:
> INVITE (cseq X)
< 403 Unauthorized (cseq X)
> INVITE+AUTH (cseq Y)
< ACK (cseq Y)
< TRYING (cseq
Hi
how to get all the users list that available in asterisk,
analog,sip,iax etc.Any cli command or AMI actions available to get this.
thanks
Nikhil
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sir,
is there any idea for this whenever 667and668 extension will dial isd call
before connect agent will dial password like ..
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cros
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.b
On Thursday 14 Jul 2011, mahesh katta wrote:
> Hi all,
>
> I have n no. of extensions in my dialer. from 456 to 556 extensions. I was
> created 2 other extensions 667 and 668 I need to allow only STD calls to
> go from this extensions.
> These all extensions are same context . I need to define th
Have something like this with necessary changes as per your requirement.
[default]
exten => _X.,1,ExecIf($[$["${CALLERID(num)}" = "667"] | $["${CALLERID(num)}"
=
"667"]]?Goto(isd,${EXTEN},1):Goto(local,${EXTEN},1))
[local]
; Mumbai Mobile Numbers
exten => _9X,1,AGI(agi://127.0.0.1:4577/c
X!,5(ALLOW),Dial(${SIPTRUNK}/${EXTEN},,To)
>> exten => _X!,6(NOTALLOW),Hangup
>>
>>
>> NoOp("Local/8600062@default-9932,1", "00") in new
> stack
>
> -- Executing AGI("Local/8600062@default-9932,1", "agi://
&g
gi://127.0.0.1:4577/call_log completed, returning
0
-- Executing MixMonitor("Local/8600062@default-9932,1",
"/var/spool/asterisk/astrec/20110714-12-00-971559566768-1310627013.4231.gsm|av(0)V(0)")
in new
stack
Jul 14 12:33:33 WARNING[17506]: ast_expr2.fl:183 ast_yyer
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