Hi,
If you're interested in what's happening in Berlin at Asterisk Tag for
the next few days, you can look here:
http://www.scribblelive.com/Thread.aspx?Id=815
I will try to keep an event trail going and notes if interviews are
posted. Anyone can post questions and I don't think there's even a
I aam in South Jersey would love to participate
On 5/24/08, | dave cantera | <[EMAIL PROTECTED]> wrote:
> dean,
> I am an active member of AUG NYC... you can email me off list for any info
> you need.
>
> also, I am preparing to start a south jersey * UG. the phila group is
> waning...
>
> thanks
Hi I better answer my own post.
I went to the code and the issue is in q931.c
/* wait for a RELEASE so that sufficient time has passed
for the inband audio to be heard */
if (c->progressmask & PRI_PROG_INBAND_AVAILABLE)
break;
Changing this lin
scenario is incoming calls on ZAP (TE410p euro isdn) to SIP (or any
other channel) and call is answered.
When I hangup on the ISDN side on the 1.2 then the SIP hangs up to
immidiatly so everything is fine (se short pri debug below).
When I do the same on 1.4.20 then it take more than 30 second
On Sat, Apr 26, 2008 at 7:13 PM, Brian J. Murrell <[EMAIL PROTECTED]>
wrote:
> On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote:
> > Does anyone have a script for manual wardialer for asterisk? not sure
> > if "wardialer" is the correct term but basically I want to call X
> > numbe
I would say anyone in the USA that is on the donotcall.gov list should
also be excluded -- unless you either qualify as an exemption or like
paying $10k for each violation.
Brian J. Murrell wrote:
> On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote:
>> Does anyone have a script for
On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote:
> Does anyone have a script for manual wardialer for asterisk? not sure
> if "wardialer" is the correct term but basically I want to call X
> number say 555- through 555-0050 and be able to listen to each
> call and when I hang
dean,
I am an active member of AUG NYC... you can email me off list for any
info you need.
also, I am preparing to start a south jersey * UG. the phila group is
waning...
thanks,
daveC
Dean Collins wrote:
This is an email to all New
York
based Asterisk users.
F
Tilghman and Jay,
Thanks for the licensing advice. If anyone is interested in replicate,
I'm now ready to distribute it under the GPL.
Regards,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
___
-- Bandwidth and
Shaun schrieb:
> Hi All,
>
> This is puzzling me greatly.
>
> The setup: PAP2T over ADSL registers to Asterisk 1.4?using SIP. Attached to
> Asterisk are SIP clients. Codec throughout G729 (only have 1 license on
> Asterisk server loaded though). When calling the SIP clients from PAP2T I
> ca
Dear Tzafrir,
Thank you for the kind response. Will do further searching, but my initial
findings are that very little is available on the net regarding the actual
ports.
Its on a public ip hosted at an isp and have full control over machine.
There is one other port that needs to be opened to han
Hi All,
This is puzzling me greatly.
The setup: PAP2T over ADSL registers to Asterisk 1.4?using SIP. Attached to
Asterisk are SIP clients. Codec throughout G729 (only have 1 license on
Asterisk server loaded though). When calling the SIP clients from PAP2T I can't
hear them but they can hear
I get this once a week myself.
Original Message
Subject: Re: [asterisk-users] excessive bounces???
From: "Michael Graves" <[EMAIL PROTECTED]>
Date: Sat, May 24, 2008 11:32 am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED].com>
I got this also.
I got this also.
In fact, it happens quite a bit for me. My mail host does not relay
email in foreign languages, which can generate a bounce. When someone
posts to the list in some asian (non roman character set) language I
always get bumped.
Michael
On Sat, 24 May 2008 16:18:29 +0100, Steve Ho
>I can make outbound calls, but when I call any of my did's they ring busy.
>A tcpdump at the Asterisk server shows no inbound traffic and neither does sip
>set debug
>show any activity. I have the providers routing set to sip user, I am using
>that user in my registration.
>
>Anyone know if ther
I got it too. I wouldn't worry.
-Original Message-
From: "Doug Lytle" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: 24/05/08 02:52 PM
Subject: [asterisk-users] excessive bounces???
My membership has been disabled for excessive bounces? Are we h
Hey Steve,
Steve Totaro schrieb:
> Darn, it was 87% off just yesterday!
If you're interested I could forward some offers for
Rolex watches, cheap software etc. :-P
Grüße,
Philipp Kempgen
--
Asterisk-Tag.org 2008, 26.-27. Mai -> http://www.asterisk-tag.org
Amooma GmbH - Bachstr. 126 - 56566
Ciao Roand
I think you should buy a book and do some reading to build up your
knowledge.
but in the meantime try something like this in the dialplan
(extensions.conf)
exten => PSTN,1,Answer() ; Answer inbound calls or internal miss-dials
exten => PSTN,2,Playback(silence/1)
exten => PSTN,3
Steve Totaro wrote:
Darn, it was 87% off just yesterday!
But with all that wonderful value-adding spam, it's worth paying more
for isn't it?
(then again, I guess that very much depends on /what/ you're paying for!)
___
-- Bandwidth and Colocatio
Another case of Extreme Asterrhea.
http://lists.digium.com/pipermail/asterisk-users/2008-May/212427.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al Baker
Sent: May 24, 2008 4:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subjec
Welcome to the club.
http://lists.digium.com/pipermail/asterisk-users/2008-May/212281.html ->
Another similar issue.
http://bugs.digium.com/view.php?id=12709 -> Bug report for it.
Mark.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carles Pina i
Estany
Al,
Yup, it seems so. Unanswered problems.
I've started a bug report, if you're interested:
http://bugs.digium.com/view.php?id=12709
Mark.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al Baker
Sent: May 24, 2008 4:35 AM
To: Asterisk Users Mailing Lis
I can make outbound calls, but when I call any of my did's they ring busy.
A tcpdump at the Asterisk server shows no inbound traffic and neither does sip
set debug
show any activity. I have the providers routing set to sip user, I am using
that user in my registration.
Anyone know if there is an
Doug Lytle wrote:
> listgolden`
>
Guess I should have done some clean up. It was time to change the
password anyways.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
___
My membership has been disabled for excessive bounces? Are we having
issues again with the list? I've check both my primary and secondary
MTA and show no issues with mail.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve
On Sat, 24 May 2008 02:29:30 -0700 (PDT), ronald ramos wrote
> hi all,
>
> we recently bought a clone box, motherboard with ICH7R raid controller (which
> i thought was a hardware raid controller). but recently i learned that those
> things are called FRAID( Fake RAID) which is basically a softw
The first thing to do is type "sip debug" on the console and place the
call from the Sipura. If you get a bunch of SIP messages flashing down
your console you know the call is reaching Asterisk and it's most
likely going to be an issue authenticating the call or a problem in
your dial plan.
If no
The main thing I have noticed over the years that causes temporary
asterisk hangs is the DNS server asterisk is trying to use becoming
inaccessible. If a network issue causes Asterisk to be unable to
connect to the DNS server you will get the classic freeze on the
console and then when the DNS serv
Darn, it was 87% off just yesterday!
On Sat, May 24, 2008 at 8:53 AM, VIAGRA (R) Official Site
wrote:
> About this mailing:
> You are receiving this e-mail because you subscribed to MSN Featured Offers.
> Microsoft respects your privacy. If you do not wish to receive this MSN
> Featured Offers e-
Barry Miller wrote:
> On Sat, May 24, 2008 at 12:01:50AM -0400, sean darcy wrote:
>> Barry Miller wrote:
>>> On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =>_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)} >
140] ? ${MAINSTUB}
On Sat, May 24, 2008 at 6:53 AM, wassim darwish <[EMAIL PROTECTED]> wrote:
>
> Hi:
> Iam an Asterisk user and i have a Sangoma A200 with 4 fxo modules and i want
> to buy Digium card with 4 fxo modules and insert it on the PCI besides the
> sangoma card ,so i will have 8 fxo channels on my aster
Hello,
(Note: again, I'm asking for experiences/suggestions, because
we have a problem in some environment that it's quite difficult to
debug, test, etc.)
I have seen, during last year, that some Asterisk has hangup up. I mean,
not crashing, we could access to Asterisk console, but phones couldn
wassim darwish wrote:
> Hi:
> Iam an Asterisk user and i have a Sangoma A200 with 4 fxo modules and i want
> to buy Digium card with 4 fxo modules and insert it on the PCI besides the
> sangoma card ,so i will have 8 fxo channels on my asterisk box ,Is that right?
> Does Asterisk make errors if
Hi:
Iam an Asterisk user and i have a Sangoma A200 with 4 fxo modules and i want
to buy Digium card with 4 fxo modules and insert it on the PCI besides the
sangoma card ,so i will have 8 fxo channels on my asterisk box ,Is that right?
Does Asterisk make errors if there is two different cards ?
There will be a slight of delay on writing files but not really a performace
issue at all.
You will hardly notice.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ronald ramos
Sent: Saturday, May 24, 2008 5:30 PM
To: Asterisk Users Mailing List - Non-
Hello all,
ive got the following setup currently:
__Sipura 3102-PSTN
|
Lan |
|
|__asterisk
i configured both asterisk and pstn to be able to receive/make calls through
each other using sip of course..
but the thing is i want asterisk that when it receives an inc
hi all,
we recently bought a clone box, motherboard with ICH7R raid controller (which i
thought was a hardware raid controller). but recently i learned that those
things are called FRAID( Fake RAID) which is basically a software raid also. so
i decide to just use Software RAID (using CentOS 5.1
Quote "
Oh and also, in my implementation there are no queues. It seems to be
>> not related, I've had it in EVERY version of Asterisk I've used."
Hmmm- maybe this should be mentioned in the next "is * Really Good Thread ?"
Mark Hamilton wrote:
> Same here.
>
> -Original Message-
> Fro
quote "And hackers ignoring pleasantries to get right down to the
technical issues isn't abusive at all"
ABUSIVE - No not at all.
Unnecessarily rude, insensitive, tacky - Yep
Jay R. Ashworth wrote:
> On Fri, May 23, 2008 at 01:25:43PM -0400, Donny Kavanagh wrote:
>
>> This is getting downright
On Sat, May 24, 2008 at 12:01:50AM -0400, sean darcy wrote:
> Barry Miller wrote:
> > On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
> >> This doesn't work:
> >>
> >> exten =>_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)} >
> >> 140] ? ${MAINSTUB}${CALLERID(num)} : ${MA
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