Hi,
I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new
Aastra SIP phones can be auto-provisioned when config files are stored in a
specific TFTP subdirectory instead of TFTP root directory.
For instance, TFTP root directory is /srv/tftp.
When config files are stored in /srv
2009/6/11 Olivier
> Hi,
>
> I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new
> Aastra SIP phones can be auto-provisioned when config files are stored in a
> specific TFTP subdirectory instead of TFTP root directory.
>
> For instance, TFTP root directory is /srv/tftp.
> Wh
Hello,
In my dialplan, I do s,n,DIAL(
)
If my called phone response and after hangup, asterisk execute the h,1,
But, if I the caller hangup at ringing (cancel), it dont execute the h,1,
Know you why?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement
On Wed, 10 Jun 2009, Alex Samad wrote:
> Hi
>
> recently bought a soekris net5501 and a tdm410 to place in there.
>
> I am having some issues attaching 12V power to the card via the molex
> card - basically the box for the motherboard is too small.
I know this mighs sound odd, but do you really n
On Wed, Jun 10, 2009 at 05:02:35PM -0700, bilal ghayyad wrote:
>
> Hi All;
>
> In dahdi: what we use instead of ztcfg -vv (that is existed /sbin/ztcfg -vv).
> ?
http://docs.tzafrir.org.il/dahdi-tools/#_dahdi_tools
(This is taken verbatim from the file UPDATE.txt in the source tarball
of dahdi-t
On 11 Jun 2009, at 08:59, BERGANZ François wrote:
> In my dialplan, I do s,n,DIAL(…)
> If my called phone response and after hangup, asterisk execute the
> h,1,…
>
> But, if I the caller hangup at ringing (cancel), it don’t execute
> the h,1,…
>
>
> Know you why?
Because the call was cance
On Thu, Jun 11, 2009 at 09:02:37AM +0100, Gordon Henderson wrote:
> On Wed, 10 Jun 2009, Alex Samad wrote:
>
> > Hi
> >
> > recently bought a soekris net5501 and a tdm410 to place in there.
> >
> > I am having some issues attaching 12V power to the card via the molex
> > card - basically the box f
I found a bug in asterisk 1.6:
http://lists.digium.com/pipermail/asterisk-dev/2009-April/037684.html
in fact, the h,1 in the macro don’t work with cancel!
Cordialement,
BERGANZ François
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-Message d'origine-
De : asterisk-u
2009/6/11 Grygoriy Dobrovolskyy
>
>
> 2009/6/11 Olivier
>
> Hi,
>>
>> I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new
>> Aastra SIP phones can be auto-provisioned when config files are stored in a
>> specific TFTP subdirectory instead of TFTP root directory.
>>
>> For i
The Asterisk Development Team is pleased to announce the second release
candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc2 is available for
immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release fixes some issues reported by the community since the first release
ca
How about this :
if you add the option 'g' in your Dial()-command, then when the caller
hangs up Asterisk will continue to execute the commands hat follow.
You could then read the ${DIALSTATUS}-variable (which will be 'CANCEL')
and execute a command based on this.
GoToIf($["${DIALSTATUS}"="ANSWER
Hello all, I have just joined this list and I'm currently working with
asterisk 1.4.17 using RealTime. Not quite sure the level of queries you
get but hopefully I'll be able to help with some input as well as
questions of my own.
Now to my first query. I'm changing the hold music on our system
Many CNET users have Cisco 3810's working with Asterisk 1.2, 1.4 and
even 1.6 as well as Astlinux versions.
Some are even using the T1 card to connect to a channel bank with number
of calls limited to the 6 DSP card in the 3810.
Contact users on the CNET VOIP list for details, but you will need:
On Thursday 11 June 2009 03:59:01 Steve Howes wrote:
> On 11 Jun 2009, at 08:59, BERGANZ François wrote:
> > In my dialplan, I do s,n,DIAL(…)
> > If my called phone response and after hangup, asterisk execute the
> > h,1,…
> >
> > But, if I the caller hangup at ringing (cancel), it don’t execute
>
Tilghman Lesher wrote:
> On Thursday 11 June 2009 03:59:01 Steve Howes wrote:
>
>> On 11 Jun 2009, at 08:59, BERGANZ François wrote:
>>
>>> In my dialplan, I do s,n,DIAL(…)
>>> If my called phone response and after hangup, asterisk execute the
>>> h,1,…
>>>
>>> But, if I the caller hangup
Anthony wrote:
> Tilghman Lesher wrote:
>
>> On Thursday 11 June 2009 03:59:01 Steve Howes wrote:
>>
>>
>>> On 11 Jun 2009, at 08:59, BERGANZ François wrote:
>>>
>>>
In my dialplan, I do s,n,DIAL(…)
If my called phone response and after hangup, asterisk execute t
Olivier schrieb:
> I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new
> Aastra SIP phones can be auto-provisioned when config files are stored in a
> specific TFTP subdirectory instead of TFTP root directory.
>
> For instance, TFTP root directory is /srv/tftp.
> When config
We had another dahdi problem this morning and got a dahdi_test -v. When
calling into Span 2 it was just dead air. There was also an HDLC Abort
at the start of the problem this time as well, but not the very first
time it happened.
Thanks.
Allan
#dahdi_test -v
Opened pseudo dahdi interface, me
David Backeberg wrote:
> I would ask the question the other way around. Are there any plans for
> Microsoft to release a unified messaging product that will comply with
> SIP over UDP?
>
>
>
I do see your point in a potential (ok who are kidding - real) risk of a
system crash with using MS ha
On Wed, 2009-06-10 at 23:00 +0100, Wayne wrote:
> I was wondering what the current development plans / patches etc are to
> allow Asterisk to talk to Exchange 2007 Unified Messaging with respect
> to adding SIP over TCP support?
There is experimental support for SIP over TCP in Asterisk 1.6.0 an
Right now, my organization is using a commercial service (OneCallNow.com),
that gives telephone notifications to all numbers in a predefined list.
Example:
-Admin records a voice message
-Service calls each number in the list, and plays the message back to them
It's a pretty handy service, albeit
Nerdvittles.com has a nice example of this, when they are up. They used it
for Phone trees for a school or something like that. Took less than 30
minutes to put in my dialplan and use.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Not too hard to do,
you can have a script generate a list of call files which automatically
ring the callers in the list and play a message
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Cheers Duncan
Christopher Stamper wrote:
> Right now, my organization is using a comme
At 02:01 PM 6/10/2009, you wrote:
> > http://www.cyberguys.com/product-search/?keyword=molex
>
>doesn't look like it, really need a 90 degree plug and I am in OZ not
>usa so postage is going to kill me
I'd buy a standard one, pull the pins, cut off the wire end of the
plug, put it back in bend th
Hi
Running 1.2.26 BRI stuffed. Calls made via PSTN via ISDN interface (Junghanns).
SIP ports mapped through firewall as we often connect from outside, but all SIP
accounts have good passwords.
However our telecoms provider picked up a few suspicious calls to places we do
not normally call at t
On Thu, Jun 11, 2009 at 3:30 PM, Paul Redstone wrote:
> Hi
>
> Running 1.2.26 BRI stuffed. Calls made via PSTN via ISDN interface
> (Junghanns).
>
> SIP ports mapped through firewall as we often connect from outside, but all
> SIP accounts have good passwords.
>
> However our telecoms provider pi
I can only suggest the most obvious cause without knowing how its
configured, sorry.
Take a look at the default context in sip.conf
for me:
[general]
context=default
my "default" context doesn't exist, so if a call comes in from an
unknown user, asterisk complains about not matching whatever numb
Still no luck getting this to work. I have been looking at the
CallManager Logs but so far that is worse then useless. Anyone out
there have any luck connecting Asterisk 1.4 and Cisco CallManager
3.3(5)?
Jimmy Ezell
http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html
___
On Thu, Jun 11, 2009 at 11:14:47AM -0700, Ira wrote:
> At 02:01 PM 6/10/2009, you wrote:
> > > http://www.cyberguys.com/product-search/?keyword=molex
> >
> >doesn't look like it, really need a 90 degree plug and I am in OZ not
> >usa so postage is going to kill me
>
> I'd buy a standard one, pull
On Thu, Jun 11, 2009 at 5:04 PM, Jimmy Ezell wrote:
> Still no luck getting this to work. I have been looking at the CallManager
> Logs but so far that is worse then useless. Anyone out there have any luck
> connecting Asterisk 1.4 and Cisco CallManager 3.3(5)?
Not exactly. We have luck loading
I'm working on replacing a SoundPoint 600 with a 650. I need to merge
these two sets of digitmaps in the polycom sip.cfg file, because the 650
locks up when I try to use the digitmap from the 600. I've included the
default digitmap from a 3.1.3 RevB polycom release.
I'd like to merge these two
>>>Very few calls have been made this way, trivial cost, but it is slightly
worrying.
That's what I thought when they hacked into one of my systems, but it is
not the cost of the calls, it is the purposed of the calls you should watch
out for. The FBI contacted the owner of the PBX, and inquire
Hello,
Where can I found information about writing modules, applications and low
level interactions for asterisk?
At http://www.asterisk.org/developers I was unable to find tutorials for
doing what I mentioned above.
Thanks in advance.
Carlos.
___
--
this is possibly the best you can find:
http://www.russellbryant.net/blog/2008/06/30/how-to-write-an-asterisk-module-part-3/
On Thu, Jun 11, 2009 at 7:50 PM, Carlos Ruiz
Diaz wrote:
> Hello,
>
> Where can I found information about writing modules, applications and low
> level interactions for aste
Hi,
As of v 1.6.1.1, can anyone tell me what the current possible values for
DIALSTATUS could be? I found
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe
it is outdated since there is no FAIL or FAILED in this list.
Thanks!
Thank you! It is really useful,
On Thu, Jun 11, 2009 at 9:13 PM, Moises Silva wrote:
> this is possibly the best you can find:
>
> http://www.russellbryant.net/blog/2008/06/30/how-to-write-an-asterisk-module-part-3/
>
> On Thu, Jun 11, 2009 at 7:50 PM, Carlos Ruiz
> Diaz wrote:
> > Hello,
> >
> >
Hello users,
have been facing problems with t38 passthrough using
asterisk 1.6.0.3.
observed also that in case of SendFAX we are not having
major issues, its almost successfull.
ReceiveFAX has problems most of the time.
we have been using a ringcentral account for testing this
receivefax.
so
Dear all ,
Is there any way for fax-on-demand on TDM FXO port using asterisk ? . I am
seeking for guidence to meet this goal .
with regards,
ashik ali . m
chen...@india .
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asteris
Hi All,
I am having some problems with Asterisk on static IP and Sipura-1001 on
dynamic IP. Is there any solutions to in the Asterisk configuration or
Sipura-1001 to re-register when the router change IP dynamic IP? Thanks.
Regards,
Kengie
___
-- Band
39 matches
Mail list logo
