Hi,
I recently found someone was using one of my Asterisk servers to make
international calls via some SIP method that allowed them to bypass
authentication (running 1.4.21.1 so I'm not sure how they did this since the
major vulnerability for this was patched in 1.4.18.1). At any rate I caught
it
,
> beep,beep...".
>
> For a better solution I would recommend you to get at least your local
> prefixes and use the correct dial string with patterns. This can be achieved
> with a script.
>
>
> On Wed, Oct 29, 2008 at 6:15 PM, arkda <[EMAIL PROTECTED]> wrote:
>
would be using r a time of 60 seconds. I've been
thinking of implementing this as a temp fix, but not something I want to
leave in place.
On Wed, Oct 29, 2008 at 5:46 PM, arkda <[EMAIL PROTECTED]> wrote:
> Thanks for the reply!
>
> I've played around with R to so
is just for early
> media, you wont get that message.
>
> For more info on the Dial command see:
>
> http://www.voip-info.org/wiki-Asterisk+cmd+Dial
>
>
>
> On Tue, Oct 28, 2008 at 6:56 PM, arkda <[EMAIL PROTECTED]> wrote:
>
>> Some additional information.
>
)
This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.
Thoughts, criticism, insults all welcome!
On Tue, Oct 28, 2008 at 12:53 PM,
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area that
I haven't needed to deal with before and I was hoping someone might have
some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or not
(
That makes sense. Thanks Steve.
On Thu, May 22, 2008 at 5:29 PM, Steve Totaro <
[EMAIL PROTECTED]> wrote:
> Echocancelwhenbridged=no should work.
>
> Thanks,
> Steve Totaro
>
> On Thu, May 22, 2008 at 5:26 PM, arkda <[EMAIL PROTECTED]> wrote:
> > Maybe that
>
> Thanks,
> Steve Totaro
>
> On Thu, May 22, 2008 at 5:07 PM, arkda <[EMAIL PROTECTED]> wrote:
> > I'm using two contexts, [internal] and [external] as in the example.
> There
> > are four digit extensions for each fax machine in my [external] context
>
s,
> Steve Totaro
>
> PS. Figured I would start with DHADI now.
>
>
> On Thu, May 22, 2008 at 4:37 PM, arkda <[EMAIL PROTECTED]> wrote:
> > Thanks for your response Steve. You almost lost me when I saw DAHDI,
> that's
> > going to take some getting used
as you described in your second post.
I'm not sure what I could do differently there...?
On Thu, May 22, 2008 at 4:10 PM, Steve Totaro <
[EMAIL PROTECTED]> wrote:
> On Thu, May 22, 2008 at 4:07 PM, Steve Totaro
> <[EMAIL PROTECTED]> wrote:
> > On Thu, May 22, 200
Hi,
I've been trying to implement a fax solution using a TDM880 (8 analog ports,
FXS). I have a PRI circuit that terminates on a TE120 at the same server.
The server is running Asterisk 1.4.18.
The idea is to have seven fax machines, each with a different number,
connected directly to the TDM880.
I've been asked to look at deploying Asterisk in a high availability
environment and I've been looking so I've been searching for methods to
decouple the voice PRI circuits from the Asterisk server so failover to
another server could take place.
I've been looking at the RedFone foneBRIDGE2 2e1 pro
Sure Shaun, I'll give it a shot. I'll contact you directly to let you know
the results.
On Wed, Feb 27, 2008 at 10:33 AM, Shaun Ruffell <[EMAIL PROTECTED]> wrote:
> arkda wrote:
> > Nothing in the console aside from what I've posted. When a DTMF tone is
> >
Just a heads up, the echo cancellation problem disappeared with Asterisk
1.4.15, zaptel 1.4.8, and libpri 1.4.3.
Still having other problems with the TE120P, but all OT from echo
cancellation.
On Mon, Feb 25, 2008 at 7:45 PM, arkda <[EMAIL PROTECTED]> wrote:
> Sorry, 1.4. Keep forgetti
You can interrogate the SIP information for some of this using the SIP debug
command on the CLI along with the udptl debug command. It's not perfect but
it works for what you're looking for.
On Tue, Feb 26, 2008 at 3:21 PM, Robert Moskowitz <[EMAIL PROTECTED]>
wrote:
> I am running Trixbox 2.4 wh
newer than .15 on Asterisk.
On Mon, Feb 25, 2008 at 7:23 PM, Kevin P. Fleming <[EMAIL PROTECTED]>
wrote:
> arkda wrote:
>
> > Asterisk revision 104093, zaptel revision 3849, libpri revision 529 all
> > from svn.
>
> Revisions of what branches? We need to know the URL
ne is
played.
On Mon, Feb 25, 2008 at 4:03 PM, Kevin P. Fleming <[EMAIL PROTECTED]>
wrote:
> arkda wrote:
>
> > [Feb 25 12:54:01] WARNING[8661]: chan_zap.c:1437 zt_enable_ec: Unable to
> > enable echo cancellation on channel 1 (Argument list too long)
>
> Can you te
, Tzafrir Cohen <[EMAIL PROTECTED]>
wrote:
> On Mon, Feb 25, 2008 at 03:27:07PM -0500, arkda wrote:
> > Hi,
> >
> > I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and
> > I've ran into an issue. After a call placed any DTMF tone causes th
; you are running the latest of each.
>
> Tzafrir Cohen wrote:
> > On Mon, Feb 25, 2008 at 03:27:07PM -0500, arkda wrote:
> >> Hi,
> >>
> >> I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9and
> >> I've ran into an issue. Aft
Hi,
I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and
I've ran into an issue. After a call placed any DTMF tone causes the server
to lock up entirely. Calls placed work just fine (except for a problem with
echo cancellation). The phone registered to the server is a Linksys S
Hi,
I recently installed a TE120P in my lab with a full voice PRI (23 channels +
1 D channel). Everything is working well except echo cancellation; for the
most part this isn't an issue unless one of the users is in a conference.
I'm getting the following error when a call is picked up (incoming
The one thing I've noticed with the latest firmware I've tried (5.2.5) is
that the speaker phone quality is improved quite a bit (full duplex) but
it's very quiet, even turned all the way up.
On Sat, Feb 23, 2008 at 8:35 AM, Jared Smith <[EMAIL PROTECTED]> wrote:
> On Sat, 2008-02-23 at 10:09 +11
Aha, thanks guys. I like the idea of just using NoOp extensions that don't
actually do anything so I'm going to give that a shot.
On Sun, Feb 24, 2008 at 2:14 PM, JR Richardson <[EMAIL PROTECTED]>
wrote:
>
>
> JR Richardson
>
> You can't map the [internal] context in dundi.conf because you have
Hi,
I'm having difficulties with using DUNDi between two servers. If it were
three I think I could control looping by limiting TTL, but with two I'm not
sure how to prevent a loop causing bad things to happen. I've tried ttl=1
but things still blow up.
The DUNDi configurations are pretty simple a
I'm a huge fan of the Linksys SPA-942s for users. They run around $125, are
pretty straightforward to manage via TFTP, and work really well with
Asterisk.
On Thu, Feb 21, 2008 at 4:07 AM, Michael J. Liberatore <
[EMAIL PROTECTED]> wrote:
> A while back i had asked about possible replacements for
Hi,
Is anyone still using backticks on 1.4? Or is there another way to pull a
variable from a shell script into Asterisk 1.4?
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asterisk-users mailing list
To UNSUBSCRIBE or update
Disabling ACPI might provide some interesting results as well. Core Duo
systems will drop to one processor. I've seen some very odd timing problems
with VMWare products on servers with ACPI turned on in the past.
On Feb 2, 2008 2:58 PM, Benny Amorsen <[EMAIL PROTECTED]> wrote:
> Johansson Olle E
Sorry if this was repeated, but I think the list is acting up and not
accepting some emails so I wanted to resend it just in case.
On Jan 29, 2008 10:15 AM, arkda <[EMAIL PROTECTED]> wrote:
> Recently with Asterisk 1.4.17 I've been running into some stability
> issues. I starte
Recently with Asterisk 1.4.17 I've been running into some stability issues.
I started looking through my logs, and I found this:
[Jan 29 09:41:45] WARNING[13132]: loader.c:620 inspect_module: Module
'codec_g729a.so' was not compiled against a recent version of Asterisk and
may cause instability.
Details on the TFTP configs can be found in the 'IP Phone Administrator
Guide' here:
http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1139435695017&pagename=Linksys%2FCommon%2FVisitorWrapper
On Jan 26, 2008 3:17 PM, arkda <[EMAIL
The SPA942s configure very readily through TFTP. You can utilize DHCP to
point to your TFTP server (option 150 I think), and the phones look for a
file called spa.cfg.
The TFTP config files are pretty straightforward and easy to configure. I
usually just keep a spa-template.cfg and the individual
I've ran into an issue (1.4.17) where anything in features.conf is being
totally ignored after the * or # for a particular feature.
Everything works just fine as long as I restrict the digit to only a * or #,
but apps that require #1 or *1 simply never get recognized.
No clue what's causing this,
I recently went through the same thing. My company was paying huge amounts
of money for voice PRIs at several locations, ongoing PBX support to a third
party, and huge amounts of money for a teleconference bridge to yet another
third party. I was bored one weekend so I implemented Asterisk. Before
Hah, thanks guys. I had tried that previously, but my syntax was off.
On Jan 22, 2008 6:19 PM, Chris Bagnall <[EMAIL PROTECTED]> wrote:
> VoicemailMain(${CALLERID(number)}) is probably what you want.
>
> Regards,
>
> Chris
> --
> C.M. Bagnall, Director, Minotaur I.T. Limited
> For full contact de
My guess is you want the server to call the user and play the voicemail?
On Jan 14, 2008 1:37 PM, Gilberto Nunes <[EMAIL PROTECTED]> wrote:
> A Monday 14 January 2008 16:25:15, Steve Johnson escreveu:
>
> Yeah! I'm just do this right now!
>
> But I want more!
>
> How can I create some extension t
Hi,
Is it possible to dial voicemail from a particular phone line and
automatically enter the extension that is being dialed from, thereby only
prompting for the password?
I've been searching around to find if this is possible, but I haven't been
able to find an example of this. I have a feeling
Hi,
I've recently come across LDAPget (version 2.0rc1) and I've been trying to
get it functional in my test environment (Asterisk 1.4.15 and MS Active
Directory 2003) but I can't seem to get it working.
I put together a test extension to try to change the CALLERID(name) by way
of a LDAP query to
There's an example here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
On Dec 12, 2007 12:56 PM, Michael Munger <[EMAIL PROTECTED]>
wrote:
> I looked in the voip-info.org wiki, and it mentions paging and intercom,
> but searching the page for "paging" or "intercom" produces give
That's exactly the information I'm looking for, thanks Noah.
On Dec 11, 2007 10:16 AM, Noah Miller <[EMAIL PROTECTED]> wrote:
> Hi -
>
> > I guess that's my question. Is this the standard method of doing faxing?
> > Just point the PRI DIDs to a TDM and hang fax machines off of the ports?
>
> I've
I guess that's my question. Is this the standard method of doing faxing?
Just point the PRI DIDs to a TDM and hang fax machines off of the ports?
If this is the generally accepted method of doing things (ie, it works very
reliably) then I'm good to go. The utilization of 3102s on the links instead
Hi,
I'm putting together a fax solution for my company that utilizes T.38. I
wanted to throw out my plan and get some feedback if anyone is doing
something similar or sees a blatant problem with it.
We're currently rolling out SPA-942 phones for the standard desk phone with
vanilla Asterisk 1.4.1
Knew there must have been an easier way and something I was missing. Thanks
Anthony.
On 10/28/07, Anthony Messina <[EMAIL PROTECTED]> wrote:
>
> On Saturday 27 October 2007 11:19:05 pm arkda wrote:
> > I've been looking around for an example of a method of reading back a
&
I've been looking around for an example of a method of reading back a caller
ID value, but I haven't found anything that doesn't use Festival. I'd rather
not resort to the Mr. Roboto voice if I can avoid it.
Playback of the numbers one at a time is perfectly fine, so I'd like to use
the default fe
> On 8/8/07, arkda <[EMAIL PROTECTED]> wrote:
>
> > I've been digging around and I haven't found a way to do this, but I
> > have a feeling I'll feel like an idiot because it's something I'm over
> > looking.
> >
> > Normally if
I've been digging around and I haven't found a way to do this, but I have a
feeling I'll feel like an idiot because it's something I'm over looking.
Normally if I need to specify an additional option (such as different
language sound files) or I'm building an Asterisk server with a lean
configurat
Nice solution Eric, thanks. Very elegant.
On 6/27/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
Matthew Brothers wrote:
>> Hi,
>>
>> I have been looking for an example of accomplishing this, but
>> I've been unable to locate something similar to what I'm trying
>> to do.
>>
>> Here's th
Great examples Matthew, really appreciate it. This is exactly what I've been
searching for!
On 6/26/07, Matthew Brothers <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I have been looking for an example of accomplishing this, but
> I've been unable to locate something similar to what I'm trying
> to do.
Hi,
I have been looking for an example of accomplishing this, but I've been
unable to locate something similar to what I'm trying to do.
Here's the scenario:
Users caller ID is set to their internal extension (200-250). This is set in
sip.conf for each user. Each user has a local DID as well (h
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