On 2/02/2017, at 9:52 pm, A J Stiles wrote:
>
> but in simple solidarity with everyone who has ever
> been pissed off by a machine-initiated spam marketing phone call at an
> inappropriate moment, I am not going to tell you how to do it.
>
Hat-tip to you, AJ :)
Pete
smime.p7s
Description:
Hello,
I operate an Asterisk server (v11.13.1) on Debian stable, and it's
rock-solid. The other day, however, I accidentally upgraded the
kernel from the stable 3.16.0 to 4.9.0. Subsequently, audio stopped
working.
Below you can find my analysis while running the 4.9.0 kernel. 888
is a simply Ech
Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My
extensions.conf file was mostly copied from server running Asterisk 1.8.
That being said! If I dial a number and get a busy signal I get the
following error:
-- SIP/voipeer-084b redirecting info has changed, p
On Thu, Feb 2, 2017 at 4:06 AM, Zakir Mahomedy wrote:
> Yes, from_user was set, removing those entries solved the problem.
>
> Can someone please explain to me the correct use for fromuser field?
>
from_user forces the user portion of the From header to a specific value on
calls that go TO the d
On Thu, Feb 2, 2017, at 08:11 AM, Saint Michael wrote:
> if a PJSIP call fails, how can I capture SIP code, like 503,603 etc?
> in old SIP channel, we had ${HASH(SIP_CAUSE,)}
> but in PJSIP it has to be the outbound channel, which is gone when the
> control returns to the calling channel.
This fun
if a PJSIP call fails, how can I capture SIP code, like 503,603 etc?
in old SIP channel, we had ${HASH(SIP_CAUSE,)}
but in PJSIP it has to be the outbound channel, which is gone when the
control returns to the calling channel.
--
Yes, from_user was set, removing those entries solved the problem.
Can someone please explain to me the correct use for fromuser field?
thanksZakir
On Wednesday, February 1, 2017 8:00 PM,
"asterisk-users-requ...@lists.digium.com"
wrote:
Send asterisk-users mailing list submissions to
hi,
i have similar problem to
https://issues.asterisk.org/jira/browse/ASTERISK-25806
do you know about some workarounds/patches for better scalability?
thanks
marek
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On Thursday 02 Feb 2017, Amelye Chatila wrote:
> Hi,
> I need to make calls to a list of numbers one at a time and once the user
> pick the phone connects to an IVR where I can get few data, after a call
> finishes the 2nd number get called and so forth.
>
> I'm familiar with Asterisk/Elastix but
To be more clear I need unattended campaign
On Feb 2, 2017 11:26 AM, "Amelye Chatila" wrote:
> Hi,
>
>
> I need to make calls to a list of numbers one at a time and once the user
> pick the phone connects to an IVR where I can get few data, after a call
> finishes the 2nd number get called and s
Hi,
I need to make calls to a list of numbers one at a time and once the user
pick the phone connects to an IVR where I can get few data, after a call
finishes the 2nd number get called and so forth.
I'm familiar with Asterisk/Elastix but the Campaign feature on Elastix does
not seem to fill th
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