Net::DNS::Async is a fire-and-forget asynchronous DNS helper. That is, the
user application adds DNS questions to the helper, and the callback will be
called at some point in the future without further intervention from the user
application. The application need not handle selects, timeouts,
8 jan 2010 kl. 08.01 skrev Tilghman Lesher:
On Thursday 07 January 2010 21:17:52 JR Richardson wrote:
On Thu, 7 Jan 2010, Tilghman Lesher wrote:
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits
hi,
i use $AGI-wait_for_digit($timeout) to wait for the user press key 1
,and then to do something.
but how can i get the return number ?
is that use $key = $AGI-wait_for_digit($timeout)
and $key will be 200 result=49 if i pressed number 1?
Thanks!
--
Best regards,
Sucan
On 8 Jan 2010, at 09:14, Zhang Shukun wrote:
i use $AGI-wait_for_digit($timeout) to wait for the user press key 1
,and then to do something.
but how can i get the return number ?
is that use $key = $AGI-wait_for_digit($timeout)
and $key will be 200 result=49 if i pressed number 1?
On 8 Jan 2010, at 02:28, John Novack wrote:
Careful, or Steve will un top post YOU!
I like it in the past. Leave me alone ;)
S
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Different Steve!!
Steve Howes wrote:
On 8 Jan 2010, at 02:28, John Novack wrote:
Careful, or Steve will un top post YOU!
I like it in the past. Leave me alone ;)
S
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This is what I was using at the time:
asterisk-1.4.21.2
libpri-1.4.6
wanpipe-3.2.7
zaptel-1.4.11
spandsp 0.0.4pre16
unknown rx_fax version.
As I can see there is a 0.0.6pre16 version now..
At the time, I used this tutorial I found on the net to setup rxfax/spandsp:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, January 07, 2010 10:30 PM
What about:
1) Fixing the slow responding DNS server?
2) Tweaking /etc/resolv.conf options?
On 8 Jan 2010, at 13:52, John Novack wrote:
Steve Howes wrote:
On 8 Jan 2010, at 02:28, John Novack wrote:
Careful, or Steve will un top post YOU!
I like it in the past. Leave me alone ;)
Different Steve!!
I agree with him though :P
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On Fri, Jan 8, 2010 at 8:59 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
This is what I was using at the time:
asterisk-1.4.21.2
I really, really prefer the faxing in 1.6. It's so nice to configure
compared to 1.4. I'll leave it to the ChangeLog and anybody else
http://www.pcworld.com/article/186308/magicjack_harnesses_femtocell_for_voip.html
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About what?
On Fri, Jan 8, 2010 at 3:39 PM, Steve Howes steve-li...@geekinter.net wrote:
On 8 Jan 2010, at 13:52, John Novack wrote:
Steve Howes wrote:
On 8 Jan 2010, at 02:28, John Novack wrote:
Careful, or Steve will un top post YOU!
I like it in the past. Leave me alone ;)
Different
Hello,
In about one hour we should be chatting with Tim Behrins of Voxbone
about their initiative, iNum. I say should because he's the
scheduled guest, but I haven't heard from him today :)
Next week, we'll be Hacking VoIP
Feel free to top post your answers, it seems to stimulate conversation.
On 8 Jan 2010, at 16:03, Randy R wrote:
On Fri, Jan 8, 2010 at 3:39 PM, Steve Howes steve-li...@geekinter.net wrote:
On 8 Jan 2010, at 13:52, John Novack wrote:
Steve Howes wrote:
On 8 Jan 2010, at 02:28, John Novack wrote:
Careful, or Steve will un top post YOU!
I like it in the past.
I would have read your message but I couldn't find it amongst all of this
garbage...
:)
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne
Sent: Friday, January 08, 2010 11:10 AM
To: Asterisk
Hello everyone,
I'm trying to turn up a SIP trunk with a Cisco UCM (Unified
Communications Manager/Call Manager). It's currently configured for
3rd party call control (3pcc). The INVITEs show up without an SDP...
Neither the Cisco admin nor myself can find any documentation on how
to
On Thu, 7 Jan 2010, David Gibbons wrote:
Yes, gmail DOES default to top posting, because bottom posting is silly
(in general, but especially for a client that hides quoted text (like
gmail)). Top posting is modern. And better. And doesn't make me scroll
through 10 thousand messages and
I have been doing this (whatever that is), since about 1976, involving
many facets, including posting on #1 CBBS out of Illinois, usenet in the
90s, and more.
It is not possible to get people to follow all the RFC rules and customs
much less the -- CR sigdashes.
There are a lot of relative
Rick Green wrote:
On Thu, 7 Jan 2010, David Gibbons wrote:
Yes, gmail DOES default to top posting, because bottom posting is silly
(in general, but especially for a client that hides quoted text (like
gmail)). Top posting is modern. And better. And doesn't make me scroll
through 10
And how will we ever re-write the 10+-year-old RFCs which no longer hold
relevance to modern email clients if nobody goes against the grain and does
what makes sense rather than what has been generally accepted?
-Dave
snip
And to add on to this: aside from whether you think it is silly or not,
On Fri, Jan 8, 2010 at 5:25 PM, David Gibbons d...@videon-central.com wrote:
I would have read your message but I couldn't find it amongst all of this
garbage...
Funny I saw your right away :)
Ok, all kidding aside, I really don't care where people post if only
they'd clip all the garbage
Hello srinivas,
I have the same issue on my Asterisk installation (Asterisk 1.4.25).
Today i've updated both res_fax.so and res_fax_digium to latest version but
no success.
pfunitasbh*CLI fax show version
FAX For Asterisk Components:
Applications: 1.4_1.1.6
Digium FAX Driver: 1.4_1.1.6
On Fri, Jan 8, 2010 at 1:47 PM, Daniel Araujo redsna...@gmail.com wrote:
I have the same issue on my Asterisk installation (Asterisk 1.4.25).
As you can see, the T38 module isn't enabled on my installation. Tried ask
google how to make it work, but found no hints yet.
Anyone can help us?
If
Rick Green wrote:
problem also becomes apparent, and that is the failure of many MUAs to
honor 'sigdashes', which is the convention of preceeding your sigfile with
a line that is 'dash dash space CR'. A compliant MUA will strip that
line and everything after it when quoting for a reply or
And some of the most rabid bottom posters are as guilty of snipping out
all the mailing list garbage added to every message. Scrolling through 5
of these to find some comment is as, or more annoying that top posting.
Randy R wrote:
On Fri, Jan 8, 2010 at 5:25 PM, David Gibbons
]
queue_log = yes
queue_log_name = queue_log
Thanks,
Best regards!!
Cristian Arguello.
__ Información de ESET NOD32 Antivirus, versión de la base de firmas de
virus 4755 (20100108) __
ESET NOD32 Antivirus ha comprobado este mensaje.
http://www.eset.com
On a well set up system you should be able to send or receive those
pages all day. If you can't, you probably have timing issues in your
Asterisk setup.
This is a uncleared question. What does timing issue exactly mean?
1) Enable internal timing and use one of the res_timing_*.so (with
In version prior to 1.6, timing is very critical for faxing, and the use of
a timing source improves fax sending/receiving., and if no timing source was
used, then you would use zt_dummy, but I am not sure how reliable that is or
was..
And from what I am reading, v1.6 is far better with faxing,
yeah, but what about internal_timing = yes in asterisk.conf
yes or no for faxing? Or is this option irrelevant for app_fax/spandsp ?
2010/1/8 William Stillwell (Lists) william.stillwell-li...@ablebody.net:
In version prior to 1.6, timing is very critical for faxing, and the use of
a timing
See this thread:
http://lists.digium.com/pipermail/asterisk-dev/2006-April/019756.html
you still need the ztdummy if no hardware timer is not available, and from
what I read, internal_timing=yes tells it use the hardware timer if
available.
-Original Message-
From:
HI Guys,
I am trying to use the RTPPage application on asterisk 1.4 using the Snom
320's?? My goal is to do the paging using a multicast IP address.
I tried the app_rtppage.c and i can only do unicast on the snom's and i was
unable to do a multicast.
Hello All.
I would like to know what codec is being used during a call. For example if I
have 3 channels on 3 active calls how can I find what codec is beeing used by
each client?
Thanks in advanced.
--
_
--
What about:
1) Fixing the slow responding DNS server?
2) Tweaking /etc/resolv.conf options?
3) Setting up a caching name server on your Asterisk host?
4) Adding the AGI server host name and IP address to /etc/hosts?
5) Using the IP address of the AGI server in your dialplan?
Ok, I
On Fri, 8 Jan 2010, Landy Landy wrote:
I would like to know what codec is being used during a call. For example
if I have 3 channels on 3 active calls how can I find what codec is
beeing used by each client?
How about something like:
asterisk -r -x sip show channel
Hi!
I am trying to use the RTPPage application on asterisk 1.4 using the
Snom 320's??
Are you asking us if you are trying to do this? Only you would know. ;-)
i have the same IP/Port to be listened on for multicast traffic on the
Snom 320's. But when i make a call to 1234, the snom 320
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