Hello,
is there an option to log calldate / start in GMT / UTC?
CSV has an option usegmtime=yes.
Best Regards
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On 08.11.20 14:18, John Fawcett wrote:
On 06/11/2020 14:28, basti wrote:
Hello,
i try to connect my SIP Client (linphone) via VPN to FreePBX.
The routing looks OK. I can ping the Endpoints and traffic is routing.
I can also Register my Sip Client.
debpbx*CLI> pjsip list contacts
Cont
Hello,
i try to connect my SIP Client (linphone) via VPN to FreePBX.
The routing looks OK. I can ping the Endpoints and traffic is routing.
I can also Register my Sip Client.
debpbx*CLI> pjsip list contacts
Contact:
Hello,
i try to setup asterisk with hylafax:
the config is:
egrep -v "(^#|^$)" /etc/hylafax/config.ttyIAX0
CountryCode:49
AreaCode: xxx
FAXNumber: +49
LongDistancePrefix: 0
InternationalPrefix:00
DialStringRules:etc/dialrules
Hello,
i try to use mail2fax with asterisk.
all i have found is old stuff like
- https://sourceforge.net/projects/asterfax/files/production/NoojeeFax or
- https://github.com/siddolo/sidfax
so I try sidfax (aka email2fax).
Now asterisk show in sip debug mode the following error.
[17-06-2020
Hello I use Asterisk 13 with FreePBX.
When I try to connect my Softphone via VPN to Asterisk I'm registered
and It's show via "pjsip list contacts"
Then I try to call an internal number / other extension I get the
following: "SIP/2.0 401 Unauthorized".
The VPN net is list in
I have try to register. no confirmation mail received. A new
registration fails with "mail address" in use.
On 12.09.19 08:03, Michael Maier wrote:
> On 11.09.19 at 15:24 Joshua C. Colp wrote:
>> On Wed, Sep 11, 2019, at 10:18 AM, basti wrote:
>>> Hallo,
>>> i
Hallo,
is there a Freepbx mailinglist? or can this be posted here?
Best Regards,
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Hello,
is there a way to view the call log (call history) from extern via
browser, XML or whatever?
At the moment I see no numbers in call-log.
Asterisk do the CIDLokkup at the moment, and the Ring Groups has an CID
Name Prefix.
So an a Phone you can see:
called Depart: CallerName and CallerID
/230>
>
>
>
> Em sex, 15 de fev de 2019 às 20:14, basti <mailto:mailingl...@unix-solution.de>> escreveu:
>
> Hello when I set qualify = yes on trunk I can't do outgoing call.
> Incoming is always working.
>
> [Feb 15 23:01:41] WARNING[12909][
Hello when I set qualify = yes on trunk I can't do outgoing call.
Incoming is always working.
[Feb 15 23:01:41] WARNING[12909][C-0012]: app_dial.c:2525
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
but my linphone is registered all the time.
when set
Hello again,
my Grandstream GXP1610 is show the call on display but no "ringing" is
hearing. The caller hear a ring tone but the phone is like muted.
With an other phone all is fine so i think thats a phone setting.
Has anyone a solution for this? I have fixed in the past but can't
remember how.
resses in the Dial() command.
>
>
> Mitch
>
> On 2/6/19 8:16 AM, basti wrote:
>> In other words.
>>
>> I there a way that both phones are ring with only one extension?
>>
>> On 06.02.19 15:05, basti wrote:
>>> both phones are in the same net.
In other words.
I there a way that both phones are ring with only one extension?
On 06.02.19 15:05, basti wrote:
> both phones are in the same net.
> when the soft phone is shut down, on hardware phone only an led is
> flashing to show an incoming call but no sound.
>
> both phon
13:54:44, Mark Wiater wrote:
>
>> These two phones are not using the same extension, are they?
>
> If you shut down the softphone, does the hardware phone then ring?
>
>
> Antony.
>
>> On 2/6/2019 8:49 AM, basti wrote:
>>> both phones are registere
ich phone vendor do you want to connect? can you make
> outgoing calls with hardwarephone?
>
> BR Cyril
>
> Am Mittwoch, den 06.02.2019, 13:00 +0100 schrieb basti:
>> Hello,
>>
>> I have some user that had have a hardwarephone and an softphone. I
>> use
&g
Hello,
I have some user that had have a hardwarephone and an softphone. I use
pjsip driver and set "Max Contacts = 2" to have register both at the
same time.
But Only the softphone is ring. the hardware phone is mute.
How can i fix this?
--
Hello,
my Asterisk is installed on my router. From my ISP I only get an dynamic IP.
In sip.conf I have try:
externhost=host1.mydns.unix-solution.de
externrefresh=300
but after reconnect I cant call from "outside".
asterisk*CLI> sip show registry
Hostdnsmgr
1.19 11:40, Antony Stone wrote:
> On Thursday 31 January 2019 at 11:36:05, basti wrote:
>
>> With softphone I mean linphone csipsimple or whatever.
>
> I know what you mean by "a softphone"; I just wasn't sure how you were
> calling
> your softphone and wh
With softphone I mean linphone csipsimple or whatever.
How should a dialplan lokks like?
On 31.01.19 11:26, Antony Stone wrote:
> On Thursday 31 January 2019 at 10:59:01, basti wrote:
>
>> Hello I use this dial paln:
>>
>> [o2-in]
>> exten => o2,1,Answer
>
Hello I use this dial paln:
[o2-in]
exten => o2,1,Answer
exten => o2,n,Playback(hello-world)
exten => o2,n,Ringing
exten => o2,n,Dial(SIP/10/20/s@no-op,25,rt)
exten => o2,n,Playtones(425/1000,0/4000)
exten => o2,n,Wait(30)
exten => o2,n,Hangup()
All is fine. Hello world is Playback and I hear a
Hello,
At the moment we have a Swyx phone server.
We would like to switch to a free asterisk PBX.
All phone extensions work as expected except the fax.
This is a analog Fax behind a Audiocode MP-112 FXS.
I have create a extension for the MP112.
The MP112 can register to asterisk.
When I send a
ote on 12/14/2017 09:36:06 AM:
>
>> From: "basti" <mailingl...@unix-solution.de>
>> To: asterisk-users@lists.digium.com
>> Date: 12/14/2017 09:36 AM
>> Subject: Re: [asterisk-users] Rewrite Outgoing Number
>> Sent by: asterisk-users-boun...@lists
On 14.12.2017 16:30, basti wrote:
Hello,
I am new on asterisk and do some tests on freepbx.
I have 2 SIP provider:
Provider1: In-/Out- Flatrate, only 1 Number
Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers
On Asterisk site i have 3 phones
(branch ??, don't know how
)
Is it possible to do something like:
Phone 1: Incoming Call: Number1/Provider1 Outgoing Call: Number1/Provider1
Phone 2: Incoming Call: Number1/Provider2 Outgoing Call: Number1/Provider1
Phone 3: Incoming Call: Number2/Provider2 Outgoing Call: Number1/Provider1
Best Regards,
Basti
I know but this is not my sole decision.
On 12.12.2017 16:17, Ron Wheeler wrote:
> If your phone system goes down and you can not get it back up until
> tomorrow afternoon because your support person is on another project,
> you may wish you had an SLA.
--
ith a
> genius in charge that you may only be able to reach after hours or a
> shop with techs of various skill levels that can give you a believable SLA.
>
> Ron
>
> On 11/12/2017 3:53 PM, basti wrote:
>> Hello,
>>
>> we plan to move a PBX to asterisk and sear
.
Best Regards,
basti
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https
Hi , i've a problem with the dispatcher module of Openser, for the load
balancing for asterisk
The schema of the network is this :
Firewall (public ip:
199.199.199.199:5060)
|
I've a question, if I have a bridged call , when the called hangup, it is
possible that the channel of the caller remain active and then execute
something (make another dial or something else)?
How hook the callee hangup ?
Thanks
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g option.
Is that what you're looking for ?
Mathieu
Antonio Basti a écrit :
I've a question, if I have a bridged call , when the called hangup, it is
possible that the channel of the caller remain active and then execute
something (make another dial or something else)?
How hook
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