It might work for you to branch on ${DIALSTRING} just after your Dial
command, if you want to handle a BUSY, NOANSWER, or other result. But if
the peer of that Dial hungup, then based on what Joshua said, it seems
there's no recovery.
--
Please point me to samples of popping and wiping hangup handlers. I don't
need to use the values returned; I just need to clear any handlers before I
push a new one.
It's not clear at
https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers+Specification how
to provide vars on the right-hand
We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to
10.0.0.228. But sometimes another of our servers becomes listed as a SIP
agent, even though the server's IP address isn't part of our sip.conf,
extensions.conf, nor any other config I know of. For example in the log
snippet
Response below...
On Fri, Dec 27, 2019 at 12:02 PM David P wrote:
>
> >
> > I'm looking for a way of detecting in my dialplan when a peer becomes
> > non-responsive after answering. [deleted] Is there a way to configure
> > a handler for this state?
> >
> &
I'm looking for a way of detecting in my dialplan when a peer becomes
non-responsive after answering. It seems that Asterisk knows when the peer
becomes non-responsive because it logs "Remote UNIX connection
disconnected" around the same time, and it seems that
if there is no following "Remote
an Auto Dialer software for ITSP
> <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition>
> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
> http://www.ictbroadcast.com/
>
>
> On Fri, Oct 5, 201
We tried to use the CURL fn to POST json, but it's sent as form data and
there seems no support for changing the Content-Type header. We switched to
invoking curl in the shell.
All the documentation I could find says there is just one parameter for the
url and an optional second for POST body. Is
Thanks for sharing this, Alex. It sounds like TURN, as a media repeater,
wouldn't work if the media must be secured (via SRTP). Is that right?
On Wed, 3 Oct 2018, 3:17 pm alex epshteyn, wrote:
> WebRTC requires a few specific things to be in place. We have blog posts
> that talk about WebRTC
, I get "Sorry, there was an error
authorizing your account. Perhaps you did not approve authorization?" I've
never received an email asking to verify my address, if that's what this
error means. I just tried re-registering, too.
On Sun, Jun 17, 2018 at 7:25 PM David P wrote:
> I al
I also just tried adding this:
same => n,Set(SIP_CODEC_INBOUND=g722)
On Sat, Jun 16, 2018 at 4:36 PM David P wrote:
> We want to record inbound channels at 16kHz, but send only 8kHz to our
> peers. I've set our default profile in sip.conf to disallow all but g722,
> and the peers
We want to record inbound channels at 16kHz, but send only 8kHz to our
peers. I've set our default profile in sip.conf to disallow all but g722,
and the peers disallow all but ulaw. We have a proxy in front of Asterisk
that is configured to disallow all but G722 also.
My test calls show inbound
FYI, we found that our peers don't hangup properly. But we would still like
to know how to get the peer's hangup handler to fire upon peer hangup,
because right now it corrupts our globals by firing after the caller's
hangup handler.
On Tue, Jun 5, 2018 at 5:40 PM, David P wrote:
> FWIW
ose by extensions. No difference.
On Tue, Jun 5, 2018 at 3:17 PM, David P wrote:
> This has been super-helpful, Eric. However, the handleHangupByPeer priorities
> below are still not run when the peer hangs-up. The last line in the cli
> when the peer hangs-up is still:
> Stri
This has been super-helpful, Eric. However, the handleHangupByPeer priorities
below are still not run when the peer hangs-up. The last line in the cli
when the peer hangs-up is still:
Strict RTP learning complete - Locking on source address
(Although sometimes there is also: Retransmission timeout
ed replacing the Dial above with:
same => n,Dial(${DialForPeer},,g)
Cheers,
David
On Tue, Jun 5, 2018 at 7:38 AM, Eric Wieling wrote:
> Use hangup handlers, they work around the issues with the 'h' extension.
>
> On 06/05/2018 05:33 AM, David P wrote:
>
>> Thanks, Anthony.
&
.
Finally, is there a way to reset all globals, maybe as a variant of
"dialplan reload"?
On Tue, Jun 5, 2018 at 1:21 AM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> On Tuesday 05 June 2018 at 08:33:26, David P wrote:
>
> > We're using Asterisk 14.7.6 and
We're using Asterisk 14.7.6 and I have a dialplan that ends like this:
same => n,Dial(SIP/${EXTEN:0:4}@peer1)
same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH})
same => n,Hangup()
When peer1 hangsup, the priorities after the Dial are executed fine. But
when the caller hangsup during the Dial, the
We're using Asterisk 14.7.6, and we're able to route particular extensions
to a corresponding automated member via SIP. Each extension has its own
specific content, and all the automated members are configured to handle
any of these extensions. We would like to achieve some scaling by putting
all
We would like to use 20-char extension values that use dashes and alphanums
after the first four digits. In order to handle these via pattern-matching,
how can I define a pattern that allows dashes? There seems to be no option
at http://the-asterisk-book.com/1.6/einleitung-regex.html#re
19 matches
Mail list logo
