Only two days until the Astricon pre-conference activities start on
Tuesday, and three days until the main conference opens up in
Phoenix! Many of the Digium staff are on-site already, preparing the
dCAP testing facilities, getting the network ready, and sorting
through the arrangements to ma
Hi,
I have inherited some code that appears to implement a kluge-y way of adding
and removing extensions, sip devices, and sip registrations dynamically.
Yep, you guessed it - it modifies the extensions.conf and sip.conf files,
and then execute script to ask Asterisk to reload the dialplan and th
Greetings to all !
I will be flying on Monday with BA from London to Phoenix so i was
wondering if anyone else is on the same plane so there will more than
inflight movies to pass the time :)
If so please contact me off-list and we can arrange to meet.
--
Stelios S. Koroneos
Digital OPSiS - Em
Luis, you can join asterisk-r2 mailing list in the same way you joined
asterisk-users, just go to http://lists.digium.com/ and select
asterisk-r2 mailing list, there you just need to provide your e-mail
address.
Moy
On Sat, Sep 20, 2008 at 11:48 AM, Luis Morales <[EMAIL PROTECTED]> wrote:
> Moy,
DesRae Mason wrote:
> accounts. Can anyone provide me with some thoughts on how to do this
> as easily and efficiently as possible?
Generate a listing of email addresses from your database, record your
broadcast message and email it to them?
Doug
--
Ben Franklin quote:
"Those who would gi
I have asterisk 1.4 set up, using MySQL, with about 5,000 voicemail
accounts. All calls to accounts are for leaving voicemail only, which is
then emailed to the users. No voicemail is kept on the system, or in a db.
I am interested in sending a pre-recorded broadcast message to all
accounts. Can
"Cisco obviously didn't buy jabber.com engineers to implement a Cisco IM
platform for their retail clients and that they must have something much
bigger in mind."
Dean, I'm right there with you. My money is on them using it as the first
step in a larger strategy to provide a framework for applicat
Moy,
How i can do to join asterisk-r2 list ? My congratulations about your
article in digium blog http://blogs.digium.com/page/2/
I will collaborate in your project and give support from Venezuela.
Regards,
Luis Morales
On Sat, Sep 20, 2008 at 7:47 PM, Moises Silva <[EMAIL PROTECTED]> wrote:
>
On Sat, Sep 20, 2008 at 12:18:42PM -0400, Dean Collins wrote:
>No I know they just bought the company and not the protocol basically
>they bought engineering bums on seats.
>[1]http://deancollinsblog.blogspot.com/2008/09/cisco-acquires-jabber.ht
>ml
>Cisco obviously didn't buy
No I know they just bought the company and not the protocol basically
they bought engineering bums on seats.
http://deancollinsblog.blogspot.com/2008/09/cisco-acquires-jabber.html
Cisco obviously didn't buy jabber.com engineers to implement a Cisco IM
platform for their retail clients and that
Is it possible to send callwaiting callerid to a channel without
actually having a call waiting, I'm thinking if messaging a handset
through the manger interface for example?
Robb
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
> I wonder what this means in the long run for the open development of this
> platform?
Not a darn thing, unless Cisco screws around and makes an incompatible
version of a jabber server and client that doesn't play according to
the protocol. Microsoft Java, anybody?
We'll see how long this list s
Wow - now this interesting
http://www.techcrunch.com/2008/09/19/cisco-acquires-jabber-for-enterpris
e-im/
I wonder what this means in the long run for the open development of
this platform?
Regards,
Dean Collins
[EMAIL PROTECTED]
+1-212-203-4357 (New York)
+61-2-9016-5642 (Sydney
On Fri, Sep 19, 2008 at 5:29 AM, <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I have a SIP request which comes from an Asterisk and which has to
> re-enter in the same Asterisk (during the same session), but during the
> second passage in Asterisk, it send me a 482 Loop Detected. So is it a
> bug or Aster
On Fri, Sep 19, 2008 at 12:54:58PM -0700, George Williams wrote:
> Hi,
>
> I am issuing CLI commands via script, using the "asterisk -rx" method.
>
> Its working great. Now, I need to get the results of the command to look
> for error messages, etc.
>
> I've tried setting several "-v" flags as
Hi!
> > specific SIP headers. Besides "wrong number", I would especially like to
> > send 302 temp moved with a specified address to deflect certain calls.
> > Is there any way to send a specific reply out of the dialplan?
>
> No. The dial plan does not provide such low-level access to the SIP
>
Hi,
I have the following symptoms:
Call X-lite / Nokia E51
X-lite press hold: Nokia DOES hear MOH
Nokia press hold: X-lite does NOT hear MOH
Call X-lite / SCCP phone
MOH works as supposed
Call SCCP phone / Nokia E51
SCCP press hold: Nokia DOES hear MOH
Nokia press hold: X-lite does NOT hear MOH
Greetings to all !
I will be flying on Monday with BA from London to Phoenix so i was
wondering if anyone else is on the same plane so there will more than
inflight movies to pass the time :)
If so please contact me off-list and we can arrange to meet.
--
Stelios S. Koroneos
Digital OPSiS - Emb
Hi Hitesh,
Usually, subscribing to DID provider is a one way thing, they can call
you to that number, but you cannot call out via that number.
If you already have a pots line available, which means you are probably
paying monthly for it already, might as well buy an fxo card and make
use of th
Stefan Gofferje wrote:
> Hi,
>
> when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong
> number" to unwelcome callers.
> Meanwhile, I am only using SIP providers (no PSTN lines any more) and I
> would like to do similar, i.e. send specific SIP headers. Besides "wrong
> number",
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