Hi all,
Lately, I've been seeing more and more instances where I get a flood of warning
messages like this:
[Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining
The next thing I know, my server is dropping calls and starting to misbehave.
I use fax via T.38, so I can't just turn
Hi,
Which version of asterisk is this !? I had the same situation in which
asterisk just consumes UDP ports and don't release the ports on call hangup
hence this error appears after a while. So I just upgraded asterisk version
and everything worked better than expected.
Regards,
Sammy
On Fri,
Ok understood. The signaling wont be a problem, but not the same with rtp
as it uses randomly ports. The idea is to have an intermediary who could
delivers both ports and ping them to both sides to keep nating open on
routers, this is what i do with rtp proxy within opensips.
But in this case no
Did you asked OpenVOX for support?
El 27/04/2012 01:48, John Millican j...@millican.us escribió:
Hello,
I have an OpenVox A400E02 (2FXO) in a box running Debian 6.0.2 running
Asterisk 1.8.6.0. I have to POTS line on it from Verizon in Virginia, USA.
Whenever I place a call to one of the two
Btw, red alarms means phisical problemscheck cable first.
El 27/04/2012 10:23, Danny Dias ing.diasda...@gmail.com escribió:
Did you asked OpenVOX for support?
El 27/04/2012 01:48, John Millican j...@millican.us escribió:
Hello,
I have an OpenVox A400E02 (2FXO) in a box running Debian
Thought this deserved a name and shame!
;)
Steve
Begin forwarded message:
From: Pavel Ismailov pavel.ismai...@gmail.com
Date: 27 April 2012 06:58:07 GMT+01:00
To: steve-li...@geekinter.net
Subject: Flashphoner
Hello!
My name is Pavel Ismailov
and I`m CEO of www.flashphoner.com
Boo, and i felt so special for a few minutes this morning! :(
--
AJ
[YOUR AD HERE]
- Original Message -
From: Steven Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 27 April, 2012 9:28:18 AM
The asterisk side has to have the router ports 5060 and 1-2 forwarded
to asterisk these are the standard ports but you could cut way down on the rtp
ports in rtp.conf then you have to tell asterisk what's the external ip of
your nat and most of the times this should work today no
Thanks,
But if i open rtp ports from 1-2 how would you ping ports from both
sides to not loose rtp or having one way audio if the ports are choosen
randomly between 10.000-20.000 in every call?
The keep alive works for signalling (Asterisks sends Options to the
contact), but not for RTP.
I'm using asterisk 1.8.7.0 and fax for asterisk
I'm receiving a multi page fax using the ReceiveFax application with the
d and f options.
The fax is being written as a tiff file but only the first page of the
fax is in the tiff file
The asterisk server knows there are multiple files in the fax
On Fri, Apr 27, 2012 at 1:22 AM, Danny Dias ing.diasda...@gmail.com wrote:
Are you sure that works for you with the same environment as mine? Its
just that im trying to understand it technically (sdp headers + sip
headers) and i do not understand how the rtp will reach both phones on
Please ignore, I was using a viewing client that didn't understand the
multiple pages
On Fri, 2012-04-27 at 14:20 +0100, Ishfaq Malik wrote:
I'm using asterisk 1.8.7.0 and fax for asterisk
I'm receiving a multi page fax using the ReceiveFax application with the
d and f options.
The fax
On 04/27/2012 09:01 AM, shayne.al...@gmail.com wrote:
I am interested to know if there is any application or way to help me
for this Scenario:
When we put Callers in Q, the MOH will stop for announcements!
how if we able to increase the MOH (RT/TX) and then play any announce
with greater RX/TX
Yep, But I think that is can be done, independent to Queue app.
If we have an application which call MusicOnHold inside! maybe we can
control simultaneously playing back of files...
if we have such ability then it can be replaced by what exist inside the
Queue.
or maybe an application which
On 27/04/12 11:57 AM, shayne.al...@gmail.com wrote:
Yep, But I think that is can be done, independent to Queue app.
If we have an application which call MusicOnHold inside! maybe we can
control simultaneously playing back of files...
if we have such ability then it can be replaced by what exist
On 04/27/2012 10:57 AM, shayne.al...@gmail.com wrote:
Yep, But I think that is can be done, independent to Queue app.
If we have an application which call MusicOnHold inside! maybe we can
control simultaneously playing back of files...
if we have such ability then it can be replaced by what
Really? Me?
Oh Pavel! I would be inestimably honoured.
On 04/27/2012 01:55 AM, Pavel Ismailov wrote:
Hello!
My name is Pavel Ismailov
and I`m CEO of www.flashphoner.com project.
We noticed that you quite active in Asterisk-user
mail list, and would like to offer you buy signature
in your
so nice! it's a good idea.
i will try it...
tnx
On Fri, Apr 27, 2012 at 8:35 PM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
On 27/04/12 11:57 AM, shayne.al...@gmail.com wrote:
Yep, But I think that is can be done, independent to Queue app.
If we have an application which call
:)
sorry , cos i am not native english...
On Fri, Apr 27, 2012 at 8:38 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 04/27/2012 10:57 AM, shayne.al...@gmail.com wrote:
Yep, But I think that is can be done, independent to Queue app.
If we have an application which call MusicOnHold
On 04/25/2012 05:29 PM, Eric Wieling wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Wednesday, April 25, 2012 6:25 PM
To: asterisk-users@lists.digium.com
Subject: Re:
On 04/27/2012 01:24 PM, shayne.al...@gmail.com wrote:
congratulations @};-
It's a match made in Heaven. I have spare signature space to sell, and
Pavel wants signature space to rent! At a low introductory rate of
US$1800/word, he and I are going to make this happen...
--
Alex Balashov -
What flavor are flashphoner minties?
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Friday, April 27, 2012 12:37 PM
To: asterisk-users@lists.digium.com
Subject: Re:
On 04/27/2012 01:39 PM, Don Kelly wrote:
What flavor are flashphoner minties?
--Don
Dailing flavored. What else?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
Hello, I'm trying to build a page system using a Dell Desktop PC optiplex
170L,
My sound card is working fine under /dev/snd/
exten = s,1,Dial(Console/snd/,20,A(trek))
exten = s,2,Hangup
But won't work! I get the following error
[Apr 27 11:44:46] WARNING[2950]: chan_oss.c:377 find_desc: could
Only the premium dailing minties. The regular flashphoner ones are indebted to
a complex vanilla ice cream + pork belly + cardboard mixture...
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity and errors.
Alex Balashov - Principal
Evariste Systems LLC
235 E
On Sunday, April 29th, 2012, the Asterisk community services
listed below will be undergoing maintenance (software upgrades and
updates). The services will be shut down at approximately 9:00 PM CDT
(2:00 AM April 30th UTC), and will return no later than 10:00 PM
CDT. We apologize in advance for
I am expanding our systems and am looking for a way to track wich asterisk
server a peer is registred on. Here is an example.
4 Asterisk servers. An peer could regester on any of the three servers.
When I a peer registers I need to some how trigger an update to a master
database to tell wich
On 04/27/2012 02:28 PM, Bryant Zimmerman wrote:
I am expanding our systems and am looking for a way to track wich
asterisk server a peer is registred on. Here is an example.
4 Asterisk servers. An peer could regester on any of the three servers.
When I a peer registers I need to some how
Hi,
Looking to do video conferencing with Asterisk and after some research I
noticed there's mainly the new confbridge application in Asterisk 10 or
there's the Medooze MCU software.
I'm not sure as how they compare feature-wise. I get the impression the
video support in confbridge is
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