Hi John,
Ah ha! Excellent. That works.
Now for a further tweak, in my stdexten I set voicemail_option with
with b or u, as appropriate and use ${voicemail_option) instead of
option in the call to Voicemail below so the correct prompt is used.
Thank you!
On Thu, 2016-07-21 at 14:53 -0700, John
On Thu, Jul 21, 2016 at 6:02 PM, Chirag Desai wrote:
> Hi all,
>
> I was using 13.5 but upgraded today to 13.9 (13.10 came out a few hours
> after I upgraded).
>
> On both 13.5 and 13.9 asterisk seems to use 100% of the CPU. This usually
> happens a few hours after starting
Hi all,
I was using 13.5 but upgraded today to 13.9 (13.10 came out a few hours
after I upgraded).
On both 13.5 and 13.9 asterisk seems to use 100% of the CPU. This usually
happens a few hours after starting asterisk. A restart of asterisk gets the
CPU back down, but only for a little while.
I think you almost have it.
In your vmfwd context have a wildcard match that sends the caller back to
the originating voicemail and then define specific extensions that are
allowed to forward.
[vmfwd]
exten => _,1,Voicemail(box@context,option)
same => n,Hangup
; Andrew Ruthven
On Thu, Jul 21, 2016 at 4:18 PM, Teijo wrote:
>
>
> 21.7.2016, 20:38, Asterisk Development Team kirjoitti:
>>
>> Bugs fixed in this release:
>> ---
>> * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
>> (Reported by
Hey,
I have free calling to between DDIs and cellphones on our group plan. I
figure it'd be nice to allow staff with those cellphones to be able to
forward callers to their VoiceMail to their cellphones using the *
feature.
I have a standard extension macro that has VoiceMail support.
So far
21.7.2016, 20:38, Asterisk Development Team kirjoitti:
Bugs fixed in this release:
---
* ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
(Reported by Alexander Traud)
Now it's possible to use dtls_cipher settings such like:
The Asterisk Development Team has announced the release of Asterisk 13.10.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.10.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 11.23.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.23.0 resolves several issues reported by the
community and would have not been possible
We have a customer who does significant ConfBridge recording every day. They
are concerned about the size of the recording that will accumulate.
>From the confbridge.conf.sample file, it mentions "the default format is 8khz
>slinear"
It is possible to change that "default format" and if so,
Stefan Viljoen wrote:
Only this one trunk consistenly has this problem for all calls received over
it. The trunk provider is using sippy on their side.
What setting / config option for the particular SIP "problem trunk" have my
trunk provider changed on their side to stop Asterisk from
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