On 1 February 2012 21:26, Andrzej Nowrot anow...@interia.pl wrote:
Hi
I have noticed new behaviour of asterisk 10.0 realtime.
In 1.6 when I was using realtime:
[somecontext]
exten = someexten1..
exten = someexten2..
exten = someexten3..
exten = someexten4..
Hi all,
I was using dahdi 1.6.2.0.9 version for a long time.
We decided to upgrade to 1.6.2.22 a few days ago.
After that we started to have some problems with dahdi channels.
PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2
We have 2 PRIs between Ericsson pbx and asterisk and a sip trunk for
Hello,
ChanSpy can not be used on a Channel that is being recorded with
MixMonitor.
How can I verify if a channel which I want to spy on, is currently not
being recorded ?!
Kind regards,
Jonas.
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On Wed, 01 Feb 2012 18:47:49 -0500, James Sharp ja...@fivecats.org
wrote:
The Cisco DDR2200 that I just got from Centurylink for DSL appears to be
just that. I haven't tested the FXS ports on it yet, though.
Cisco announces the end-of-sale and end-of-life dates for the Cisco
DDR2200, DDR2201,
You may used even capturing in the case... when call is recoding in
conference
On Wed, Feb 1, 2012 at 4:04 PM, Kingsley Tart kings...@skymarket.co.ukwrote:
Hi,
I want to create a system for incoming calls where, under some
circumstances, callers get routed straight to voicemail (or some
As soon as I activate the exterip/localnet config there is no response at
all, as if that IP address desappeared.
Any packets send to it simply get no response.
I've considered being linux kernel routing issue, but since without the
exterip/localnet config it works OK I don't think this is the
Whats asterick?
I blame my spell checker! :-P
Do you have anything to offer in terms of help or advice on the
issues/questions I posted?
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New to
I think you might want to split your questions first.
this might work from local ISP network, but in my experience it might
depend on provider.
1. You can't have multiple externip, but it's not necessary to run two
Asterisk instances, because you can set routes to different destinations
via
I think you might want to split your questions first.
I thought that instead of creating a dozen different threads (and
clogging the ML in the process) it would be better to put everything
into one place - just pick the issue (or issues) you could address and
leave (i.e. delete) the rest
Hi,
I have IVR and when I press 1, asterisk calls my mobile phone.
If my mobile phone is offline, asterisk transfers to asterisk voicemail.
I'd like asterisk detects my mobile voicemail and if my mobile voicemail
answers, asterisk transfers to asterisk voicemail.
For that, I used AMD.
So I have
I currently have an Asterisk 1.8.8.1 system set up with SIP accounts
as well as a Wildcard TDM400P REV I card with both FXS and FXO
ports - FXO is connected to outside lines, FXS connected to inside
analog phones. Everything about the setup works fine except one thing
-
after making calls to
Hi Warren,
Device A is behind NAT with regards to asterisk server. As far as localnet
statement first I did configured localnet = 130.8.2.0/255.255.255.0 as per
local network, after that made a SIP call and the message I'm getting is
listed below;
[Feb 2 11:14:52] WARNING[23868]:
2012/1/31 John Knight j...@classiccitytelco.com
Personally, I don't think what Digium is doing is necessarily a perfect
approach (hey, what is? we're all human), but they've vastly improved the
quality of Asterisk from a support perspective.
I also agree that IMHO, Asterisk quality has
Hi,
I'm not sure what you mean. Can you elaborate?
Cheers,
Kingsley.
On Thu, 2012-02-02 at 18:13 +0530, virendra bhati wrote:
You may used even capturing in the case... when call is recoding in
conference
On Wed, Feb 1, 2012 at 4:04 PM, Kingsley Tart
kings...@skymarket.co.uk wrote:
Hello guys.
When I am trying to send fax through T38 to linksys SPA (properly
configured etc. - I have tried it with other systems), I'm getting error
and fax is not delivered.
I'm getting this errors in asterisk.log:
WARNING[687] udptl.c: No UDPTL ports remaining
ERROR[687] chan_sip.c: UDPTL
What happens if you do a SIP RELOAD instead of restarting Asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sent: Thursday, February 02, 2012 1:14 PM
To: asterisk-users@lists.digium.com
Good question.
I can't simulate it right now so I will try it today or so...But I'm not
expecting, that it help, because I tried couple of ways before and restart
was the only way.
But thanks anyway. Any other ideas?
What happens if you do a SIP RELOAD instead of restarting Asterisk?
Maybe you should change your values in udptl.conf? By default the range is
4000 to 4099, but is effectively 4001 to 4099 because the protocol doesn't
use even numbers by default, so it runs out of entries in 500 tries.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
well...I did so to avoid that lock again...but this is not solution...
Maybe you should change your values in udptl.conf? By default the range
is
4000 to 4099, but is effectively 4001 to 4099 because the protocol doesn't
use even numbers by default, so it runs out of entries in 500 tries.
Agreed - I think the solution is a patch to udptl.c to reset the counter
instead of dying with this message (just my opinion).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sent: Thursday, February
On 02/01/12 13:29, Christian Gansberger wrote:
Hello List!
I'm searching for SIP-Providers in the following countries:
Russia
Ukraine
Poland
I need a geographical number for each country, maybe a prepaid
SIP-Account, trunking is not important.
Has anyone some experience with these countries?
On Thu, Feb 02, 2012 at 10:45:21AM +0200, Oguzhan Kayhan wrote:
Hi all,
I was using dahdi 1.6.2.0.9 version for a long time.
We decided to upgrade to 1.6.2.22 a few days ago.
After that we started to have some problems with dahdi channels.
PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2
Hi All;
Which asterisk version that support the ability to have the configuration in
the database?
Regards
Bilal
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Tzafrir Cohen
Sent: Thursday, February 02, 2012 10:38 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk dahdi problem.
If you go to a web site called google.com, and enter "asterisk version that support the ability to have the configuration in the database", and read the first search result, you will get your answer.Christian Savinovich
Original Message
Subject: [asterisk-users] Asterisk
On Tue, Jan 31, 2012 at 12:54:41PM +, Arthur Stanfield wrote:
Hi Gilles,
You can't tunnel UDP through SSH.
For the record: you can. But it's not really a good idea. Two options:
1. ssh -D: dynamic port forwarding. Which basically means that it
creates a socks4/socks5 proxy. You can now
On Thu, 2 Feb 2012, Tzafrir Cohen wrote:
Oh, and for the record, you can tunnel practically on top of anything.
Just in case you're not familiar with it: IP over DNS (which means you
don't even need direct access, and can use proxied DNS queries).
http://code.kryo.se/iodine/
I figure you
On Tue, Jan 31, 2012 at 11:39:21AM -0600, Kevin P. Fleming wrote:
I've created a page on wiki.asterisk.org outlining some changes
we're proposing to make to the Asterisk release and support cycles.
As always, before implementing any changes of this type, we'd like
to collect some community
On Wed, Feb 01, 2012 at 06:47:49PM -0500, James Sharp wrote:
On 02/01/2012 02:17 PM, bilal ghayyad wrote:
Hi All;
I heard from some friends that there are a dsl router that has Linux OS
and it has asterisk on it, so the ip phone can register on this router,
also if the router has FXS or FXO
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