1:1 nat, I even turned off iptables.. same issue.
Guess I will try install wireshark when I get back next week, im done farting
with this tonight, when I get back from fort Lauderdale next week I will play
with it some more.
From: asterisk-users-boun...@lists.digium.com
Hi Kevin
I just found something interresting:
http://www.faqs.org/rfc/rfc3960.txt
1. Unless a 180 (Ringing) response is received, never generate
local ringing.
2. If a 180 (Ringing) has been received but there are no incoming
media packets, generate local ringing.
Thanks for reply. Any other suggestions .
On 12/20/2010 05:52 PM, Service clients - VDI CENTER wrote:
i believe there is a way to do it using asterisk and flashphoner
++
2010/12/20 Gilles codecompl...@free.fr mailto:codecompl...@free.fr
On Fri, 17 Dec 2010 15:51:33 +0530, Nikhil
Hi
Does anyone have any rough idea how far away 1.8.3 is?
We can't deploy 1.8 yet because of this issue
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
On 11.02.2011 12:37, Ishfaq Malik wrote:
Hi
Does anyone have any rough idea how far away 1.8.3 is?
We can't deploy 1.8 yet because of this issue
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
Have you tried issue18403.patch ?
--
That makes two of us. I tried asking on asterisk-dev but had no reply.
- Mensagem original -
Hi
Does anyone have any rough idea how far away 1.8.3 is?
We can't deploy 1.8 yet because of this issue
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
--
Ishfaq
2011/2/10 Tilghman Lesher tilgh...@meg.abyt.es
On Thursday 10 February 2011 06:13:38 Rodrigo Lang wrote:
I wonder if it is possible, without touching the source code, to
Asterisk save the cdr with date in unix time instead of the default
date. It's possible?
The answer is, it depends
Hello list,
I'm using realtime queues and noticing that prompts are not played as
expected.
Database :
announce =
queue_youarenext = queue_youarenext
queue_thereare = queue_thereare
queue_callswaiting = queue_callswaiting
queue_holdtime =
queue_thankyou =
queue_reporthold =
On 02/11/2011 02:11 AM, Benoit Panizzon wrote:
Hi Kevin
I just found something interresting:
http://www.faqs.org/rfc/rfc3960.txt
1. Unless a 180 (Ringing) response is received, never generate
local ringing.
2. If a 180 (Ringing) has been received but there are no
hi,
can anymore drop me a asterisl's config for digium te220b (with ec) or
at least some good tutorial of configuratin e1 line with that card ?
thanks in advance,
robert
--
_
-- Bandwidth and Colocation Provided by
Try to check your kernel version as per the wiki.sangoma.com website. I hope
that will solve your issue.
On Fri, Feb 11, 2011 at 12:17 PM, Roi Stork roi.st...@gmail.com wrote:
Trying to install wanpipe 3.5.18.
No errors during compile. But when I reach the point where wanpipe and
dahdi_cfg
Check out the Random Application and the RAND function, Here is a
quick untested example for either.
exten = s,1,Answer
exten = s,n,Background(privacy-please-stay-on-line-to-be-connected)
exten = s,n,Random(33:${CONTEXT},s,FILE1) ; 33% Num1
exten = s,n,Random(33:${CONTEXT},s,FILE2) ; 33% Num2
Hello,
only the field periodic_announce is played, but not according to the
value in periodic_announce_frequency.
If I set periodic_announce_frequency to 10, than still this announce
is played every 25 seconds.
Where does this 25 seconds come from ?
Why is queue_thankyou played, even if
I thought it has been resolved in 1.8.2 version
Thanks,
Satish
Date: Fri, 11 Feb 2011 08:46:36 -0200
From: vinic...@canall.com.br
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8.3
That makes two of us. I tried asking on asterisk-dev but had no reply.
So did I but I'm having the same problem in Asterisk 1.8.2.2 and the
asterisk issue number I pasted the link of has Target Version 1.8.3
On Fri, 2011-02-11 at 15:59 +, satish patel wrote:
I thought it has been resolved in 1.8.2 version
Thanks,
Satish
Date: Fri, 11 Feb 2011
On 11-02-11 04:37 AM, Ishfaq Malik wrote:
Does anyone have any rough idea how far away 1.8.3 is?
If you are hard up for a release, you can use the latest 1.8.3 RC[1].
[1]
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.3-rc2.tar.gz
--
Paul Belanger
Digium, Inc. | Software
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Friday, February 11, 2011 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8.3
On
On Fri, Feb 11, 2011 at 7:59 AM, satish patel satish...@hotmail.com wrote:
I thought it has been resolved in 1.8.2 version
Issue 18403 was not resolved in 1.8.2, but in 1.8.3-rc1. Release
1.8.3-rc2 was cut on 1/20/2011, so hopefully the full release will be
out soon.
You can see where the
On 11-02-11 11:14 AM, Danny Nicholas wrote:
Isn't that a little Bleeding edge? Guess that's why
I'm still using the 1.4X set.
No? OP asked for asterisk 1.8.3, currently asterisk-1.8.3-rc2 is the
latest RC. If no regressions are found with the new patches, it will
become 1.8.3.
Basically, if
William,
As I understand you have to upgrade your GTalk to Google Voice to be
able to place and receive PSTN calls in Asterisk.
The easiest way is to subscribe to GMail using your GTalk credentials.
As you are in GMail try calling PSTN from there.
Then go to Google Voice / Settings / Phones
I am running 1.8.3 and my BLF lights have stopped working. The hints appear
to be intact when I use core show hints. But none of the phones are getting
the BLF updates. This has happend in the past and I have had to restart my
server. What could be causing this to occur. It did not do this
Hello everybody,
Anybody here knows about Astmail ? I have set up in a server but something is
going wrong! i can open its web interface but when I put the extension number
and its password I receive:Invalid mailbox or password My asterisk is 1.6 and
my S.O is debian lenny.
I know this is
Hey all,
I tried to do some searching but I found snippets and I am having trouble
putting it all together.
I want to have an option off the IVR that plays back the announcement for
the day. At the end of the message, I want the caller to get kicked back to
the previous menu.
The
I have asterisk 1.8.2 in development and i can blind transfer from A to C
without any issue. Or may be i am doing wrong thing?
How do i reproduce this error ?
-S
Date: Fri, 11 Feb 2011 11:41:37 -0500
From: pabelan...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re:
Here is the patch did you apply it ?
https://issues.asterisk.org/file_download.php?file_id=28206type=bug
Date: Fri, 11 Feb 2011 08:46:36 -0200
From: vinic...@canall.com.br
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8.3
That makes two of us. I tried
ERIC HERRON wrote:
The conditions are that I want the recorder to dial a feature code
that prompts him to record the message. He then presses 1 to accept.
This gets saved as announcement.wav.
You'll need to supply some of your own audio prompts, I recorded my own,
here is the code I have:
On Fri, 11 Feb 2011, ERIC HERRON wrote:
I want to have an option off the IVR that plays back the announcement
for the day. At the end of the message, I want the caller to get kicked
back to the previous menu.
The conditions are that I want the recorder to dial a feature code that
prompts
Hi All
what does the Compiler Option mean LOTS_OF_SPANS ?
The description is: More than 32 DAHDI spans
Does this mean, more than 32 DAHDI Channels ?
I have TWO T1 line so do i need to select this option ?
-S
--
On 11-02-11 03:57 PM, satish patel wrote:
what does the Compiler Option mean LOTS_OF_SPANS ?
The description is: More than 32 DAHDI spans
Does this mean, more than 32 DAHDI Channels ?
I have TWO T1 line so do i need to select this option ?
No, your card would have 2 spans.
EG:
1 span =
Thank you so much
That means 1 card 1 span 2 card 2 span
-S
Date: Fri, 11 Feb 2011 16:04:19 -0500
From: pabelan...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk compile option DAHDI SPANS
On 11-02-11 03:57 PM, satish patel wrote:
what does the
On 11-02-11 04:15 PM, satish patel wrote:
That means 1 card 1 span 2 card 2 span
Not really, for example 1 card can have 4 spans (Digium TDM410P). You
can usually substitute the word 'port' for 'span'. EG: My card has 2 T1
ports - My card has 2 T1 spans.
--
Paul Belanger
Digium, Inc. |
Hi gang,
In 500 words or less (if possible), please explain what is a
legal music-on-hold file? My boss hates the stuff provided with the
distribution and I figure that I'm asking for trouble if I take my Les Mis
tracks and run them through Audacity and SOX to make new files.
Port 5222 opened?
Jian
On 11-02-10 09:54 PM, William Stillwell wrote:
I was getting unable to make channel..
So, this is what I am doing..
Service stop asterisk
Purge modules
Make clean
Remove all traces of iskemel
Recompile that. With , add needed entrée into ldconfig.
Verify iksemel
On 11 Feb 2011, at 22:37, Danny Nicholas wrote:
In 500 words or less (if possible), please explain what is a legal
music-on-hold file?
Depends on the country, and what licence you posses. Googling 'countryname
hold music regulations' may help.
S--
I am having problems playing files with the playback command, also with the
Dial (A()) option this is the output from console:
[Feb 12 01:58:41] WARNING[1569]: file.c:650 ast_openstream_full: File
home/abejide/Desktop/a.wav does not exist in any format[Feb 12 01:58:41]
WARNING[1569]:
On Sat, 12 Feb 2011, ayodele abejide wrote:
I am having problems playing files with the playback command, also with
the Dial (A()) option this is the output from console:
This is the dialplan:
exten = 1003,n,Playback(home/abejide/Desktop/a.wav)
Don't specify the file type. Asterisk will
On Sat, 12 Feb 2011, ayodele abejide wrote:
I am having problems playing files with the playback command...
And don't hijack other people's threads :)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com
I tried what you suggested and this is the console output:
[Feb 12 03:18:41] WARNING[2774]: file.c:650 ast_openstream_full: File
/var/lib/asterisk/sounds/home/abejide/Desktop/a does not exist in any
format[Feb 12 03:18:41] WARNING[2774]: file.c:956 ast_streamfile: Unable to
open
As the error suggest, try checking /var/log/messages for possible hints on
what went wrong.
Make sure you configured the device with wancfg_dahdi script first.
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474
On 12/02/11 04:02, Bryant Zimmerman wrote:
I am running 1.8.3 and my BLF lights have stopped working. The hints
appear to be intact when I use core show hints. But none of the phones
are getting the BLF updates. This has happend in the past and I have
had to restart my server. What could be
Dears;
I am looking for a way to handle callers via queuerules, but am not able to
exactly understand the meaning and affect of this two variables on Queue
Application, and how it change the priority of a caller to be answered
sooner.
QUEUE_MAX_PENALTY
QUEUE_MIN_PENALTY
tnx
--
Alrighty Gents, let's see if any of you have encountered this
one...Variables losing their value...I'm setting a variable with four
underscores (used to be two, had same issue) so it can be inherited by child
channels, and then the next line in the dialplan I use it but it appears to
be
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