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 recog
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, ho
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 w
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 w
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:
dtls_cipher=
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 I've
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 Alexander Traud)
>
>
> No
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
ext
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.
The
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 asterisk. A restart of
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
Hi Joshua
Thanks for the response.
Interesting that you mention that toll-free numbers can do this, this
problem trunk happens to receive calls from the national telecoms provider
here (Telkom SA) sourced from a toll-free number. The SIP trunk provider has
ported that toll free state telecoms com
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