On 8/20/2011 12:46 PM, Paul Belanger wrote:
Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound
google voice calls?
confirmed on asterisk 1.8.6.0-rc1
pre-patch behavior: ring-no-answer
post-patch behavior: expected
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony
Messina
Sent: Saturday, August 20, 2011 10:36 AM
To:
On 11-08-21 02:54 AM, Jeremy Kister wrote:
On 8/20/2011 12:46 PM, Paul Belanger wrote:
Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound
google voice calls?
confirmed on asterisk 1.8.6.0-rc1
pre-patch behavior: ring-no-answer
post-patch behavior: expected
Thanks
Version 2.0 of app_flite just got released.
Flite For Asterisk provides the Flite dialplan application, which
allows you to use the Flite TTS Engine with Asterisk.
It supports 8kHz and 16kHz sample rates to provide the best
possible sound quality along with the use of wideband codecs.
It works
Version 2.0 of app_espeak just got released.
eSpeak For Asterisk provides the Espeak dialplan application,
which allows you to use the Espeak speech synthesizer with Asterisk.
It supports the following languages:
Afrikaans, Albanian, Armenian,Cantonese, Catalan, Croatian, Czech,
Danish, Dutch,
On 08/20/2011 02:24 PM, Bruce B wrote:
What's the point of having the metrics then? They are inaccurate
and deceiving. If there is no benefit to showing the real metrics
then why not change it to Status = Reachable than showing a
number?
Because it's still more useful than not having it?
If
If the peers are SIP you could do:
akl*CLI manager show command SIPpeers
Action: SIPpeers
Synopsis: List SIP peers (text format)
Privilege: system,all
Description: Lists SIP peers in text format with details on current status.
Variables:
ActionID: idAction ID for this transaction. Will
Hello,
How to allow inbound anonymous call on asterisk ?
Sincerely,
Tseveen
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On 08/22/2011 01:38 AM, tseveendorj wrote:
How to allow inbound anonymous call on asterisk ?
allowguest = yes, in sip.conf [general] section.
However, I do not advise it.
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