On 2016-11-27 06:46 AM, Max Grobecker wrote:
Hi,
you could try switching the DTMF mode of the ATA's SIP peer (and also in the
ATA itself) to INBAND transmission.
In this mode, the ATA doesn't need to recognise DTMF tones and your Asterisk
can interpret it.
For this to work, the ATA needs to us
i'm using gammu[1] with a 3g dongle and my own chip with my preffer
provider. It can send over 700 every our and receive to. I don't know if
you need asterisk and sms in the same way but with this tool you can make
everything. It has python tools to.
Best regards.
Emiliano.
[1] https://wammu.eu
hmm. i think customer will not agree this is correct behavior
from pcap it looks like there is missing CANCEL to the second device
Dne 30/11/2016 v 19:42 Sam Basan napsal(a):
Your second call is not without sound, there is simply no call at all.
As the first answer the call his channel and t
I have played around with iaxmodem and hylafax and have a few working
installations where PRI's are involved. I have a new customer that will
be sending inbound fax calls via a new (SIP) DID provider we are working
with (yup, same one from the last message I just sent), and they support
T38.
We are trying to work with a new DID provider and I find myself
confused. Their standard integration is to send the call with no
authentication. I am expected to whitelist all their possible gateways,
and accept their calls I guess with no peer definition. I actually have
it working this w
Is there are way to specify the display name of a resource in a resource
list? I have setup a resource list in Asterisk 13 for 1234. All is working
on the device, but I want to show "Joe User" instead of "1234". Any
thoughts?
--
Derek Bolichowski wrote:
HI Michael,
You can set this in sip.conf:
session-timers=refuse
I know of this option - it doesn't help, because the provider ignores it
(on some calls) and the call is dropped anyway.
Normally, there is no problem with the timers. And the problem which
occurred he
Your second call is not without sound, there is simply no call at all.
As the first answer the call his channel and the external call channel
connected.
The second device simply off hook but his channel have no external channel
to connect.
It's looks like a simple telephony glare.
Sam
בתאריך 30
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier
Sent: Wednesday, November 30, 2016 12:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropped call aft
Hello all!
I can see a strange problem during invite in dialog in the context of
timer handling.
Given is the following incoming call from provider at 8.195.88.234 (2@2)
to my asterisk at 28.19.57.152 (1@1):
After 900s suddenly *asterisk* starts the timer reinvite - I would have
expected the rei
hi,
our customer reports problem when 2 agents answer the call in the same time
faster operator (device) answer the call, but the second is showed up
(on device) and call is without sound
asterisk 13.9/app_queue with strategy ringall/operators via Local
channel with sip device (chan_sip)
d
On Wednesday 30 Nov 2016, Michele Pinassi wrote:
>[stuff deleted]
> but on a call directed to, es. FAX_3700 i got:
>
> [Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309
> handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension
> 'FAX_3700' rejected because extension not fo
Hello,
The letter "X" is reserved for dialplan patterns. You should escape it
this way :
_FA[X]_
Best regards
Jean Aunis
Le 30/11/2016 à 11:45, Michele Pinassi a écrit :
Hi all,
my dialplan is:
/;
==//
Hi all,
my dialplan is:
/;
==//
//; FROM VOIP//
//;
==//
//
//[from-voip]//
//include => default//
//
//[default]
/
/; FAXes//
/
Yes, it works !
Thanks :-)
Michele
On 30/11/2016 10:19, Jonathan H wrote:
> I think it might be related to this?
> https://issues.asterisk.org/jira/browse/ASTERISK-26391
>
> I think I remember having to edit logger.conf - this is what mine
> looks like now:
> console => notice,warning,error
> me
I think it might be related to this?
https://issues.asterisk.org/jira/browse/ASTERISK-26391
I think I remember having to edit logger.conf - this is what mine
looks like now:
console => notice,warning,error
messages => notice,warning,error
Try that, restart asterisk and see if it works :)
On 30 N
Hi all,
after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) don't show
what's happens. I've trying setting debug and verbose to 100 but
nothing, no show. All commands works as expected but i can't what's
happens on my asterisk server.
asterisk*CLI> core show settings
PBX Core settings
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