red into
a DECT base station, that works with Asterisk?
Best regards, Sebastian Nielsen
smime.p7s
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Check out th
”tre” ( https://www.tre.se ) has in sweden, one of sweden’s largest
phone operators, they are 4th the largest phone operator. (1: Telia, 2: Tele2,
3: Telenor, 4: Tre)
Then you understand why I wonder WTF people are doing…
Best regards, Sebastian Nielsen
Från: asterisk-users-boun
an the DID provider
also give you outbound calling? Most likely, but that doesn't mean that the
best way to go is to route outbound calls via the carrier that is providing you
DIDs.
On Thu, Mar 11, 2021 at 4:34 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote:
I reallt don’t under
as company
2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from
company 1 to company 2 – then company 2 owns your DIDs.
Best regards, Sebastian Nielsen
Från: asterisk-users-boun...@lists.digium.com
För Alexander Perkins
Skickat: den 12 mars 2021 01:23
Till
It sounds like there is more of the problem that neither the agent or customer
knows when to start talking, ergo, when the call is "Connected", thus the OP
wants the agent to start talking before the customer is brought in front of
that agent.
Another solution would be to just play a "fake"
om
För Frank Vanoni
Skickat: den 7 oktober 2020 19:17
Till: asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] Anyone that know of DECT "client" for asterisk?
On Sat, 2020-10-03 at 22:25 +0200, Sebastian Nielsen wrote:
> many providers in sweden have started disabling SIP accou
high echoes in the phones.
The idea is to have something simulate a DECT handset, connect to the
provider's router, and thus be able to still use asterisk.
Best regards, Sebastian Nielsen
smime.p7s
Description: S/MIME Cryptographic Signature
--
__
>>I can see the point you're making here, but what's going to do this after 30
*minutes* of normal call?
I was more into, if there is some feature that somehow triggers after 30
minutes of call - and this feature is unsupported on some client, which causes
it to drop the call. For example, if
it turned out it
outright rejects packets with unsupported features.
Best regards, Sebastian Nielsen
-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
För C.Maj
Skickat: den 22 augusti 2020 20:03
Till: asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] Chann
Yes, this means that a provider which only provides IP-access (for example a
broadband operator), ergo, when it doesn’t terminate a call, but where the call
terminates directly at a enterprise, does not need to force the end customer to
implement call verification in their PBX.
Basically, if
You could use permit/deny in the sip.conf.
That would require your script to update sip.conf dynamically and reload the
config for each time user wants to update their accepted location.
To avoid excessive reloads, you could have that the changes will take effect
after 00:00, so you have a
, Nov 16, 2019 at 7:59 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote:
What would be the best way to solve this problem? Anyone else that have got the
same problem with Android’s native SIP client, especially on Samsung phones?
I do not know if the bug is in Android nati
16, 2019 at 7:45 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote:
Hello.
I have a problem with the native Android SIP client, not acknowledging the call.
Sent a message to the list for some weeks ago containing a sip debug log, but
it only got stuck in moderation queue due
t possible to tell Asterisk to just ignore the lack of
acknowledgement from Android somehow?
Basically, for Client sip09 (username), never hang up for the reason 18
(NO_USER_RESPONSE), threat like user response was received always.
Best regards, Sebasti
Hello.
I have the following in sip.conf
[sip09]
type=peer
defaultuser=sip09
nat=yes
qualify=no
secret=sip09
host=dynamic
context=outgoing
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=h263p
deny=0.0.0.0/0.0.0.0
permit=192.168.2.2/255.255.255.255
jbenable = yes
jbforce
Business Specialist
Digium Certified Asterisk Professional
High Powered Help, Inc.
p:
678-905-8569
w:
<https://hph.io> hph.io e: <mailto:m...@hph.io> m...@hph.io
On 3/21/19 3:01 PM, Sebastian Nielsen wrote:
How did the page system answer the call when it was used wit
How did the page system answer the call when it was used with the analog
system?
You could propably fake those signals from inside asterisk, and cause it
to answer.
Från: asterisk-users För Michael
Munger
Skickat: den 21 mars 2019 20:00
Till: asterisk-users@lists.digium.com
Ämne:
of ”Registred” to your trunk operator.
Från: Ivan Demkovitch
Skickat: den 15 november 2018 18:01
Till: Sebastian Nielsen ; 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Ämne: Re: SV: [asterisk-users] Queue not dialing out to cell phone for some
reason
Sebastian,
I don't
I would suspect that the cell phone does use battery saving causing the SIP
application to lose registration with the server. Would also suggest using TCP
with a fairly short keepalive to prevent the cellular network from tearing down
the connection to the asterisk server.
You need to go into
to
a missedcall.txt log file. (call should be logged in 3 case, but not in 1
case)
2 is easy to detect, as these always are failed (non-answered) calls.
Best regards, Sebastian Nielsen
smime.p7s
Description: S/MIME Cryptographic Signature
I found very useful info here:
https://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE
In other words, on the asterisk1 box, you need to fetch from SIPPEER in
extensions on asterisk1 box, and then populate connectedline.
SIPPEER is the callee leg of the call, and CONNECTEDLINE is the
Personally, if I was a client, I would rather have the personell answer the
phone than make a outgoing call, if I would choose.
If you think of billing and costs.
So if a client allows outgoing, I don't think they have any problems with
answering a call immediately following either.
But I assume
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Nielsen
Sent: Wednesday, May 10, 2017 2:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to detect fak
Use a callback.
So when clocking in/out, they will hear a random 4 digit PIN, like "Enter
four, three, six, eight at the callback".
After they hangup, the phone will ring, and then they will have confirm with
the 4 digit PIN.
If they arent in presence: the phone at the site will ring, and the
You need to ensure that traffic to the SIP box is sent to the correct IP. Also
if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT
and traffic redirection works as is so the Asus router knows it should send the
traffic through tunnel and not via WAN.
IMPORTANT: Then
Yes its called the state table. This because connection IP:PORT has a
relationship with inside IP 192.168.x.x port X.
I guess you have configured the redirect port to be same on both?
Eg 5070 goes to *1:5060 and 5080 goes to *2:5060
What you need to do, is to have different inside ports
Im using SMS successfully over VoIP. No problems at all. You however need to
use a good codec.
However, I don’t use the MessageSend application, instead I use the raw SMS()
application.
This works by the SMS centre calling my fixed landline from a specific number,
I detect the callerid,
Why RS485? Whats wrong with a simple 3-wire connection (monospeaker, monomic,
ground) where you short monomic to ground on button press?
Then you could use a simple usb device + device server to convert fron
"smartphone headset" to usb then to network.
On the server, you use a SIP phone
In extensions, I have this.
The variable "oex" contains the original extension called, and is used to
route outgoing calls internal or external depending on several factors.
But now, im implementing a system that should require a passcode upon
calling a "sensitive number".
Here is the
Theres always garbage in the end of the files.
I do this when I want to read a file:
same => n,Set(featurefile=/home/test/feature-1.txt)
same => n,Set(unfilteredfeat2=${FILE(${featurefile},0,1,l,u)})
same => n,Set(feature2=${SHIFT(unfilteredfeat2)})
After that, add a , inside
ailable" device is
that because its offline or not registred, then the person owning it can
obviously not be engaged in the call, and thus its wise to ring the other,
online device.
Best regards, Sebastian Nielsen
smime.p7s
Description: S/MIME Cry
I have a Asterisk set up. In this, I want to use queues.
Now I want to group "agents" into groups, such as so if one phone in a group
is busy, the whole group is considered busy.
Eg:
Group1:
SIP/Dad
SIP/DadsMobile
Group2:
SIP/Mom
SIP/MomsMobile
If there is three persons in
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