On Thu, 7 Jul 2011 09:32:08 +0500, Faisal Hanif fai...@vopium.com
wrote:
Community can help you better if you provide some details about you scenario
and requirement.
It's a very simple scenario: The Asterisk server is connected to a
VoIP provider for calls to the PSTN, and I'd like to have
On Thu, 2011-07-07 at 14:23 -0400, Jeremy Kister wrote:
On 7/7/2011 9:32 AM, Ishfaq Malik wrote:
I'm having the same issue on 1.8.3.2 (with a couple of patches)
exten = s,1,Set(CHAN=${SHELL(asterisk -rx core show channels | awk
'/^SIP\/vgw1-/ { print $1 }' | head -1)})
This
On Mon, Jul 11, 2011 at 02:29:25PM -0700, Steve Edwards wrote:
The second AGI, 'neutered-agi' is an AGI of 'production length' (around
1,600 lines) and supporting access to a MySQL database. The AGI is of
'production length' but still exits after reading the AGI environment
variables
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Tuesday, July 12, 2011 3:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Monitoring connection to VoIP provider?
On
Doesn't seem to help. I did it early yesterday morning and have
another 'stuck' call this morning
Does anyone have any other ideas on what I can do to correct this?
thanks
Shawn
CLI core show channels
Channel Location State Application(Data)
DAHDI/8-1
On Fri, Jul 08, 2011 at 10:58:06AM -0400, Shawn L wrote:
I have a situation where I have an Asterisk box which receives 8
analog lines from a
Mitel PBX and then drives 8 cordless SIP phones in a 1-to-1 mapping (a
call coming in
on port 1 of the digium FXO board is delivered to SIP phone 1, an
*
DHAVAL , can you help me in designing the same ?*
On Sat, Jul 9, 2011 at 5:50 PM, G M gm.cu...@gmail.com wrote:
Anyone has Experience ?
On Fri, Jul 8, 2011 at 2:18 PM, G M gm.cu...@gmail.com wrote:
I am using Vicidial and I am looking for someone who can help with DB
Driven IVR.
On 07/11/2011 09:48 PM, d tbsky wrote:
1. SFA can not be registered after 26 July. so I want to prepare a
backup machine for our server. I read in the document that I can
re-register my SFA once. so I want to make sure if I can re-register with
my backup server now, and in the same time my
On 07/12/2011 08:26 AM, Kevin P. Fleming wrote:
It is unknown whether it will continue to be usable after that period;
Skype has the ability to disable SFA from accessing the Skype network
if they feel that is what they want to do. Since it won't get any
updates between now and then, it is
On Tue, Jul 12, 2011 at 00:22, Philippe Sultan philippe.sul...@gmail.comwrote:
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 :
Steve Edwards wrote:
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'm not disparaging Perl programmers or the language.
Let's make this a Spider-man contest. The No-prize will be the
satisfaction of seeing how it actually works. Write stevestest.agi in Perl,
PHP and C. The program must load the AGI variables, do a MYSQL QUERY and a
MYSQL INSERT. Post your source and results using this methodology:
time for i in
On 07/12/2011 09:33 AM, Matthew J. Roth wrote:
Just think how fast Linux would boot if all of the init scripts were
rewritten in C and compiled (they probably have some pipes that could be
removed, too!!). Of course, it's pretty nice to be able to easily read and
modify them, but execution
A quick to implement open source network monitoring tool is smokeping:
http://oss.oetiker.ch/smokeping/index.en.html
Tobi Oetiker and Niko Tyni's awesome tool can monitor latency on a number of
layers, and maintains charted records of connection quality.
It has a probe specific to SIP:
Hi
Like we can define cdr field format for csv, is it possible to define if
cdrs are stored in a database?
Also, what will be size limit for database CDR storage ?
--
_
-- Bandwidth and Colocation Provided by
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup
Sent: Tuesday, July 12, 2011 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] CDRs
Hi
Like we can define cdr field format
On Tuesday 12 Jul 2011, 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
tried at this list. I hope there are SFA users or Digium people can
solve my confusion.
Poor you!
To my mind,
Hi Guys,
I have been trying to implement the following for days but with no success,
any help would be greatly appreciated
My asterisk box gets calls from the SIP interface and forwards to the DAHDI
interface for example
--sip.conf-
[smycontext]
type=friend
host=xxx
fromuser=xxx
Read the wiki / manuals
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup
Sent: Tuesday, July 12, 2011 11:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] CDRs
Hi
Like we
On Tue, Jul 12, 2011 at 10:06:12AM -0500, Kevin P. Fleming wrote:
On 07/12/2011 09:33 AM, Matthew J. Roth wrote:
Just think how fast Linux would boot if all of the init scripts were
rewritten in C and compiled (they probably have some pipes that could be
removed, too!!). Of course, it's
hi:
thanks for all the information. I don't use skype and I ban skype
at our network. but there are some people who use skype and want us to
use skype to contact them. SFA is my saver because our users can use
their phone to talk with skype users and no need to install any skype
software.
I
On Tue, 12 Jul 2011, Matthew J. Roth wrote:
Just think how fast Linux would boot if all of the init scripts were
rewritten in C and compiled (they probably have some pipes that could be
removed, too!!). Of course, it's pretty nice to be able to easily read and
modify them, but execution time
Tzafrir Cohen wrote:
Well, there are a number of separate optimizations in systemd:
1. Delayed loading of services (or even not loading them at all, if not
needed. E.g.: don't load CUPS if nobody needs it.
2. Paralelized loading of services (though there have been other
Thats my issue, i hope someone could suggest something:
Phone A - Phone B
== Using SIP RTP CoS mark 5
-- Executing [01@default:1] Dial(SIP/00-0076,
SIP/01) in new stack
== Using SIP RTP CoS mark 5
-- Called 01
-- SIP/01-0077 is ringing
--
Deployed a new server different mobo and problem went away. Same version of
asterisk, same sangoma card.
Sent from my android device.--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting mysterious
dropped calls. This only happens on calls that are outbound on Dahdi and mostly
happens in conference calls particularly 8xx-xxx-
This is the output of the hangup.
[Ksebpbx1*CLI
[0KPRI Span: 1 q931_hangup:
On Tue, 12 Jul 2011, Matthew J. Roth wrote:
I recognized the code you posted. It's mine:
Thank goodness you didn't try to embarrass me.
Thank you for acknowledging that it was not my intent.
You just used my code as an example of how a non-programmer would use
a language, called piping
So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting
mysterious dropped calls. This only happens on calls that are outbound
on Dahdi and mostly happens in conference calls particularly
8xx-xxx-
This is the output of the hangup.
[Ksebpbx1*CLI
[0KPRI Span: 1 q931_hangup:
Sent from a computer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Mark Rosedale
Sent: Tuesday, July 12, 2011 4:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Steve,
Apology accepted. As I said in the original post, I hold you in high
regard so your criticism was hard to take. I still think that the trade-
off between readability and optimization is up for debate, but it's
certainly nothing to hold a grudge over.
I can tell you one thing for
what does sip show peers say?
On Tue, Jul 12, 2011 at 12:40 PM, Matiss Jekabsons mat...@jekabsons.lv wrote:
Thats my issue, i hope someone could suggest something:
Phone A - Phone B
== Using SIP RTP CoS mark 5
-- Executing [01@default:1] Dial(SIP/00-0076, SIP/01)
in
On Mon, Jul 11, 2011 at 06:45:08PM -0700, Steve Edwards wrote:
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
On Mon, Jul 11, 2011 at 06:45:08PM -0700, Steve Edwards wrote:
while read line; do
epoch=`echo $line | cut -d '|' -f 1`
if [ $epoch -ge $start_epoch -a $epoch -le $end_epoch ]; then
echo $line
fi
done /var/log/asterisk/queue_log
[snipping snippy comments about improving the
Hi List,
I have a Asterisk + FreePbx Server setup with around 10 SIP extensions
and 1 VoIP trunk (CordiaVoIP), when we dial-out to any number call is
being dropped with the following message on asterisk log:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called
Your trunk shows busy:
* -- Called CordiaVoIP/639285010430
-- SIP/CordiaVoIP-0015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)*
Try this in the CLI (asterisk -r):
*core set verbose 0*
*sip set debug peer CordiaVoIP*
And then make a call and read why
Sorry I do not understand it, here is result after:
Audio is at 172.16.9.15 port 15022
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.211.94.211:5060:
INVITE sip:639285010...@lasip1.cordiaip.net
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