On Mon, Jul 11, 2011 at 02:17:42PM -0700, Steve Edwards wrote:
>> El 08/07/11 12:50, Steve Edwards escribió:
>
>>> *) You can execute hundreds of AGIs written in C in the time it takes
>>> to load the Perl interpreter and parse your script.
>
> On Fri, 8 Jul 2011, Miguel Molina wrote:
>
>> Just cu
On 11/07/11 23:42, Steve Edwards wrote:
'Standalone' AGIs still have advantages in lower complexity and less
impact on failure. If a bug takes out your fastagi daemon it can affect
all calls.
On Tue, 12 Jul 2011, Vincent Sweeney wrote:
I'm pretty sure if you have a bug in your AGI code it's
Will you consider alternatives such as siptosis? The uncertainties are really
there for SFA
CK Lee
On 12 Jul, 2011, at 10:48 AM, d tbsky wrote:
> hi:
> I am a SFA (skype for asterisk) user. I had ask Digium questions
> about SFA usage in the future. but they seem too busy to reply. so I
> tr
hi:
I am a SFA (skype for asterisk) user. I had ask Digium questions
about SFA usage in the future. but they seem too busy to reply. so I
tried at this list. I hope there are SFA users or Digium people can
solve my confusion.
1. SFA can not be registered after 26 July. so I want to prepare a
ba
On Mon, Jul 11, 2011 at 5:29 PM, Steve Edwards
wrote:
Many times, I've made the statement that you can execute hundreds of AGIs
written in C in the time it takes to load an interpreter and parse a script
written in PHP or Perl.
I can see I'm going to spend the rest of my days erasing 'hundre
Also they tend to be used more by 'non-programmers' who get away with
'stupid' stuff like calling out to system() and piping a bunch of
commands together because they don't know how to use the language
properly :)
On Mon, 11 Jul 2011, cbul...@gmail.com wrote:
I understand your point but I don
On 11/07/11 23:42, Steve Edwards wrote:
On 12/07/11 9:29 AM, Steve Edwards wrote:
Many times, I've made the statement that you can execute hundreds of
AGIs written in C in the time it takes to load an interpreter and
parse a script written in PHP or Perl.
Well, now that I know better, let's
On Mon, Jul 11, 2011 at 5:29 PM, Steve Edwards
wrote:
> Many times, I've made the statement that you can execute hundreds of AGIs
> written in C in the time it takes to load an interpreter and parse a script
> written in PHP or Perl.
I've truly enjoyed this thread. And while startup time is certa
Also they tend to be used more by 'non-programmers' who get away with
'stupid' stuff like calling out to system() and piping a bunch of
commands together because they don't know how to use the language
properly :)
I understand your point but I don't share it There are a lot
Asterisk-P
On 12/07/11 9:29 AM, Steve Edwards wrote:
Many times, I've made the statement that you can execute hundreds of
AGIs written in C in the time it takes to load an interpreter and parse
a script written in PHP or Perl.
Well, now that I know better, let's not perpetuate an ancient claim.
'Dozen
On 12/07/11 9:29 AM, Steve Edwards wrote:
Many times, I've made the statement that you can execute hundreds of
AGIs written in C in the time it takes to load an interpreter and parse
a script written in PHP or Perl.
It would be interesting to see the same types of tests run against
fast-agi -
The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.5.0 resolves several issues reported by the
community and would have not been possible with
Many times, I've made the statement that you can execute hundreds of AGIs
written in C in the time it takes to load an interpreter and parse a
script written in PHP or Perl.
Recently, a Doubting Thomas asked me to substantiate my claim.
I suspect nobody has made the effort to implement an AGI
The destination channel dies right after your Dial statement exits,
but you can retrieve the info in the channel that's still alive :
exten => _XXX,n,Dial(SIP/${EXTEN})
exten => _XXX,n,NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)})})
Works fine on the Asterisk server I'm running (1.8.
El 08/07/11 12:50, Steve Edwards escribió:
*) You can execute hundreds of AGIs written in C in the time it takes
to load the Perl interpreter and parse your script.
On Fri, 8 Jul 2011, Miguel Molina wrote:
Just curious... have you timed this to demonstrate?
What? You want to confuse the i
Hello,
I'm trying to figure out what was the return code of SIP for a call.
The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to
retrieve the peer name using ${CHANNEL(peername)}, I have an error message
that CHANNEL does not have peername or it is not available to be used.
I
Of course I would like to say that the problem is not solved.
regards
Le 11/07/2011 13:11, Florent THOMAS a écrit :
Satish and IchFaq, thank yor for answering so fast.
I checked the queue conf and the retry is put on zero :
/[0860]
announce-frequency=30
announce-holdtime=no
announce-position=y
Am 11.07.2011 19:11, schrieb Jerry Geis:
Is there a method to "lock" asterisk into memory
such that once its loaded it does not get paged out?
"lock" into memory: disable swapping. (But this might have other
"impacts" ;-) )
"increase probability to be in memory": Add RAM and stop other applica
Is there a method to "lock" asterisk into memory
such that once its loaded it does not get paged out?
I have ran into a couple times where it seems like asterisk needs to be
paged back into memory to start answering a call.
This is on a machine that is using the ALSA port to send audio over.
Som
That patch to 1.8 was a very simple change: modify one line, add another
line. Should be easy and straight-forward to replicate on 1.4.42. (Not using
1.4 anymore over here, otherwise I would've provided the patch.)
--
_
-- Bandwidt
On Mon, Jul 11, 2011 at 8:22 AM, Doug Lytle wrote:
> Doug Lytle wrote:
>
>> I've been searching the Jira issue tracker and found a ticket:
>>
>
> What I ended up doing was to copy the app_meetme.c out of the 1.4.30 source
> and compiled it into my current Asterisk setup. I now have PIN prompts.
I was going to ask about DMA but wasn't sure, what does the host report on
DMA and what does the guest OS report from cat /proc/dma?
Just curious if the host OS gives bogo-dma.
I am looking at timing issues which a think Free Switch has figured out
along with T38 a while ago.
I care slightly abo
I agree that call files are not an appropriate way to solve this.
I would like to move back to using the original Page() application
which had always worked for us with 1.4
My initial testing found that MOH from a streaming source such as
Shoutcast only worked if I disabled the DAHDI timing modul
Hi!
FYI: There is now an official IANA registration to map phone numbers to
IAX URIs.
http://tools.ietf.org/html/rfc6315
regards
Klaus
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Doug Lytle wrote:
I've been searching the Jira issue tracker and found a ticket:
What I ended up doing was to copy the app_meetme.c out of the 1.4.30
source and compiled it into my current Asterisk setup. I now have PIN
prompts.
Doug
--
Ben Franklin quote:
"Those who would give up Esse
Dear Lali,
I'll try your proposal ASAP.
Regards
Le 10/07/2011 10:52, Florent THOMAS a écrit :
Hy all of you,
I've successfully installed a freepbx solution with 10 extensions :
- 5 on Linksys SPA922
- 1 on Linksys SPA942
- 1 on Thomson ST022
Everything seems to work fine with all the hardpho
Satish and IchFaq, thank yor for answering so fast.
I checked the queue conf and the retry is put on zero :
/[0860]
announce-frequency=30
announce-holdtime=no
announce-position=yes
autofill=no
eventmemberstatus=no
eventwhencalled=no
joinempty=yes
leavewhenempty=no
maxlen=0
memberdelay=0
music=def
I do not see the L() option on that Dial line.
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> salaheddine elharit
> Sent: Monday, July 11, 2011 4:36 AM
> To: Asterisk Users Mailing List - Non-Commercia
the CLI show this :
-- Executing [0678922645@agents:1] Set("SIP/223-6ec45a88",
"CALLERID(number)
=520460587") in new stack
-- Executing [0678922645@agents:2] MixMonitor("SIP/223-6ec45a88",
"zap_g1_06
78922645_1310376223.93960.wav|av(0}V(0)") in new stack
== Begin MixMonitor Recording SIP/2
What have you set the retry parameter for this queue?
On Sun, 2011-07-10 at 13:04 +0200, Florent THOMAS wrote:
> Hy,
>
> I'm currently working with one queue and whatever I change in the
> config, it stills a gap of 6 seconds during which no agents are
> ringing for this queue.
> Is ther any para
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