Hi All,
suppose this:
Dial(SIP/somecarrier/somenumber/60/L(360)M(td|${EPOCH})
where 60 is the seconds to wait for the callee (the called party) to answer
L(360) is the absolute limit of the call once it has been answered, in ms
M(td|${EPOCH}) is the macro to execute when the call gets
Hi All,
on this hardware:
http://www.sun.com/servers/x64/x2200/specs.xml
would one of the following 4 ports PRI cards be ok ?
http://www.voipango.com/en/ISDNInterfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE420BF-T1/E1-PRI-PCIe-HW-EC.html
Hi All,
does Digium provide a service support for a compatibility question about their
PRI hardware ?
Thanks and have a nice day.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix,
On Mon, 7 Sep 2009 02:43:54 +0100
Ex Vito ex.vitor...@gmail.com wrote:
The system specs mention PCIe expansion slots, so your only
option is the TE420B.
--
exvito
Hi Ex Vito,
shouldn't the card be low profile ?
Thanks and have a nice day.
On Mon, 7 Sep 2009 08:48:25 -0500
Juan Cardoza jcard...@tpmex.com wrote:
Hello
What is your Asterisk problem?, may be I can help you...
I had configure a T1 Card TE121 connected with and AVAYA PBX
Best regards
Hi Juan,
thanks for your help.
I'm going to choose a 4 ports PRI digium card
Hi All,
suppose this call flow:
there are two Asterisk servers, they are connected through a IAX2 trunk.
The users use SIP.
The user A on the Asterisk server 1
calls the user B on the Asterisk server 2.
They talk for a while and then the user B does an attendant transfer to the
user C on the
Hi Danny, sorry you are correct:
Difficult to say since you don't say if you are on 1.2, 1.4 or 1.6
both Asterisk are running version 1.4.21.2
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hi All,
are you aware of any solution which can encrypt calls between a mobile gsm and
isdn (asterisk) ?
Thanks for your attention,
have a nice day.
Mike
--
_
-- Bandwidth and Colocation Provided by
Hi All,
Sunday question: does mISDN work on kernel 2.4 ?
Thanks and have a nice day,
Mike
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
Hi All,
does the Asterisk's 'Dial' command have some hooks to execute commands as soon
as the 'ringing' signal is received ?
For example: can a call be dropped 1 second after the called party's phone
started to ring ?
I'm using version 1.4.
Thanks for supporting,
have a nice day.
Mike
--
Hi All,
I would like to know if you can confirm that, if using origination via AMI, as
documented here:
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
it is not possible to set the max duration of a call.
I mean: what you would do with the L (limit) parameter of the
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
Hi Ryan, thanks for your comment.
Unfortunately
${DIALSTATUS})
exten = _allow_caller_id.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 22, 2010, at 1:31 PM, mancyb...@gmail.com wrote:
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have you tried setting
Hi All, sorry for the off topic.
I own 3 vegastream 50 BRI-s 4 ISDN ports gateways with firmware version 6
and I need some advanced parameters available only from firmware version 7.
I am sure that I need those parameters because changing the vega gateway with a
20$ cologne pci card in an
Hi All,
for a vicidial server which uses only voip,
which is the minimum telephony card which would provide the required clock
timing source for conferences to work properly ?
Maybe the Digium TDM410PLF card
without any daughter card
would do the job ?
Thank you very much for supporting.
Good news, very well.
Thank you very much and have a nice day,
Mike
On Sat, 02 Oct 2010 11:38:49 -0500
Shaun Ruffell sruff...@digium.com wrote:
On 10/2/10 11:24 AM, mancyb...@gmail.com wrote:
for a vicidial server which uses only voip, which is the minimum
telephony card which would
On Tue, 05 Oct 2010 17:30:49 +0100
Paul Hayes p...@provu.co.uk wrote:
On 02/10/10 17:24, mancyb...@gmail.com wrote:
Hi All,
for a vicidial server which uses only voip,
which is the minimum telephony card which would provide the required clock
timing source for conferences to work
Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for everything else)
Question:
how can I know if the call was not answered because of timeout or because the
callee was busy ?
(I'm using
On Wed, 19 Jan 2011 17:03:03 +0100
Thorsten Göllner t...@ovm-group.com wrote:
Am 19.01.2011 16:57, schrieb mancyb...@gmail.com:Hi All,
in an AGI script, if executing the Asterisk command Dial, I only get
result = -1 (if the call has been answered by the callee)
and
result = 0 (for
Hi All,
I'm headbanging on this from a couple of days, begging here for some help :)
I'm configuring tls on asterisk for the first time
to experiment with an open (public) service idea
about having asterisk accepting any sip user (with the sip.conf option
'autocreatepeer=yes')
and call each
20 matches
Mail list logo
