Displaying 20 results from an estimated 5000 matches similar to: "Call preemption"
2019 Nov 28
2
PJSIP device_state_busy_at, how does this work?
Hi Gang
According to:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk12Configuration_res_pjsip-endpoint_device_state_busy_at
And endpoint should return busy if this number is reached.
We have PBX Trunks registering to the Asterisk.
So we want to limit the number of concurrent calls to a PBX and return
busy, if more than the configured number of channels
2015 Jun 28
1
Branch based on call volume
?I meant how many calls are in progress on a particular trunk. (Sorry - I didn't even think of the other interpretation).
________________________________
From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Matt Riddell <lists at venturevoip.com>
Sent: Sunday, June 28, 2015 9:26 AM
To: Asterisk Users List
Subject: Re:
2016 Aug 22
3
Dial and start music on hold after timeout
Sorry, I forgot to write that the SIP peer must keep ringing while the
announcement is being played.
Le 22/08/2016 ? 17:42, John Kiniston a ?crit :
> This seems like the obvious answer but maybe I'm misunderstanding the
> question.
>
> exten => s,1,Dial(SIP/alice,20)
> same => n,Playback(myannouncement)
> same => n,NoOP(Whatever else you want to do goes
2016 Aug 23
2
Dial and start music on hold after timeout
How about:
exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40)
[delayed-announce]
exten => 555,1,Wait(20)
same => n,Playback(myannouncement,noanswer)
same => n,NoOP(Whatever else you want to do goes here)
The 'noanswer' option on the Playback means that SIP/alice should continue
to ring for the remaining 20 of the 40 seconds, as the Playback will not
answer
2016 Aug 22
2
Dial and start music on hold after timeout
Thank you for the idea. The problem with RetryDial, is that it will
cancel the first call, play the announce and then dial the SIP peer once
again, so the telephone will display a missed call. I would prefer to do
everything in a single call.
Le 22/08/2016 ? 17:57, John Kiniston a ?crit :
> You could try using RetryDial() instead of Dial, It supports playing
> an announcement.
>
2020 Jan 30
2
Need feedback on the use of AMI events generated by MESSAGE requests
On Thu, Jan 30, 2020 at 12:18 AM Jean Aunis <jean.aunis at prescom.fr> wrote:
> Hello,
>
> I use UserEvents generated by the Message/ast_message_queue channel with
> the UserEvent application.
>
> Regards
>
> Jean
>
Thanks Jean. We're looking at alternatives.
> Le 29/01/2020 à 20:31, George Joseph a écrit :
>
> For those of you who actually
2016 Aug 23
2
Dial and start music on hold after timeout
Maybe try progress() instead of answer ()
?????? 23 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???:
> Thank you, I just tried your suggestion. Strangely, the announcement is
> played only if I try to dial a SIP peer which is not available (not
> registered to be more precise). If the SIP peer is available, I only get
> the ring tone, and never hear
2016 Aug 22
2
Dial and start music on hold after timeout
Hello,
I am searching a way to dial a SIP peer, and if it does not answer
within 20 seconds, play an announcement to the caller. This means that
the caller would hear a ring tone for 20 seconds, and only then hear the
announcement if the callee did not answer.
I know it is possible to do this with ARI, but in this particular case I
do not want to use ARI. I would like to do this purely with
2017 Dec 13
2
DTMF emulation with SIP INFO and direct media
Hello,
I think there is an issue when DTMF are handled with SIP INFO and direct
media is enabled.
When I receive a SIP INFO, the logs tell me that a "DTMF begin" is
generated, but no related "DTMF end" is generated, unless the call is
ended. Here is an excerpt of the logs :
*--- SIP INFO received **on **SIP/xxx-00000004:*
[Dec 13 11:56:16] DTMF[18193][C-00000005]
2020 Jan 30
1
Need feedback on the use of AMI events generated by MESSAGE requests
On Thu, Jan 30, 2020 at 3:18 AM Jean Aunis <jean.aunis at prescom.fr> wrote:
> Hello,
>
> I use UserEvents generated by the Message/ast_message_queue channel with
> the UserEvent application.
>
Do you use any aspects of the channel itself in the user events, or merely
the contents of the user event and what you've placed in it?
--
Joshua C. Colp
Asterisk Technical Lead
2020 Jan 29
3
Need feedback on the use of AMI events generated by MESSAGE requests
For those of you who actually process SIP MESSAGE requests... Do you use
any of the AMI events generated by the "Message/ast_msg_queue" channel?
We want to change that channel to an "internal" channel that doesn't
generate AMI events (for performance reasons) but we need to know if
anyone's using them first.
Thanks!
--
George Joseph
Asterisk Software Developer
2013 Jul 19
2
Meetme and maxusers option
Hi all.
I'm trying to limit the number of participants in a conference room
with the realtime option "maxusers", but it doesn't work.
Someone have this option working properly?
--
thiagoc
"O povo n?o deveria temer o governo. O governo ? quem deveria temer o povo."
V de Vingan?a
2018 Nov 30
2
Asterisk non-root - selinux - astdb
Hi
I'm trying to use Asterisk running as non-root user and selinux enabled.
Asterisk is running ok, but astdb not works. When i try to put in astdb,
console shows this message:
WARNING[1853]: db.c:350 ast_db_put: Couldn't execute statment: SQL logic
error or missing database
CentOS 7.5.1804
Asterisk certified/13.21-cert3
[root at sv03 asterisk]# ls -lahZ /var/lib/asterisk/astdb.sqlite3
2019 Jan 11
2
[asterisk-app-dev] Multiple ChannelDestroyed events for the same channel
Hiya,
When I hang up on a call to my stasis app I’m getting multiple channelDestroyed events for the same channel:
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone
app.js:1126:13) 2019-01-11 10:28:29
app.js:985:13) Channel was destroyed:
2017 Oct 06
2
Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP
Hi
Is it a normal behavior of Asterisk put a call on hold when receive a
Session Progress with media address 0.0.0.0 in SDP? I believe the call on
hold should be initiate with a re-invite.
Thanks
--
Att,
Rafael Saraiva
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2017 Mar 09
2
Trying to get SMS from GXV3240 to trigger dialplan code.
I am trying to send SMS from my grandstream GXV3240
Asterisk receives the message in a NOTIFY block.
How can I get asterisk to run dialplan code when receiving these Notify
SMS Message Blocks.
I can then route them to my SMS provider.
Any ideas are appreciated. Below is debug of a message sent from the phone
when received no dialplan code is triggered.
I am wounding if I need to
2016 Oct 13
4
Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)
Back to basics here. I want to match on one OR two digits.
The following two both work, but ONLY for more than one digit, which
is not as expected from the docs (see below).
exten => _X.,1,SayNumber(${EXTEN})
exten => _[0-9].,1,SayNumber(${EXTEN})
This next one will ONLY match 2 digits, as expected, but the first two
SHOULD match one or more, right?
exten => _XX,1,SayNumber(${EXTEN})
2011 Dec 12
1
[LLVMdev] Preemption with LLVM
Hey all,
I'm investigating LLVM for use for a future project of mine, and I was
wondering whether something is possible. Specifically, I'm wondering
if there's a way to force preemption of a green thread-style task -
something like Erlang's "processes", where if a task executes for too
long, it is preempted. [1]
My main goal here is to avoid having to write my own
2011 Jan 18
2
Surprise Thread Preemptions
Hi,
I would like to know about which threads will be preempted by which on my OpenSolaris machine.
Therefore, I ran a multithreaded program "myprogram" with 32 threads on my 24-core Solaris machine. I make sure that each thread of my program has same priority (priority zero), so that we can reduce priority inversions (saving preemptions -- system overhead). However, I ran the following
2005 Oct 16
1
GROUP and GROUP_COUNT
I have a macro and when I call it I have something like this:
exten => s,1,NoOp(Group Count: ${GROUP_COUNT(MYGROUP)})
exten => s,n,Set(GROUP()=MYGROUP) ;Set Group
exten => s,n,NoOp(Group List: ${GROUP_LIST()})
exten => s,n,NoOp(Group Count: ${GROUP_COUNT(MYGROUP)})
The GROUP_COUNT returns zero before the call to GROUP but also returns 0 after
the call to GROUP.
If I