Le 16/02/2018 à 05:30, the...@sys-concept.com a écrit :
On 02/15/2018 04:49 PM, Joshua Colp wrote:
On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
Thanks again for the hint.
Here is the output from asterisk.
The call is coming on Audocodes gateway from: pstn-
But
On 02/15/2018 04:49 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>
>
>
>>
>> Thanks again for the hint.
>> Here is the output from asterisk.
>>
>> The call is coming on Audocodes gateway from: pstn-
>>
>> But asterisk display:
>> Found peer
Thelma
On 02/15/2018 07:16 PM, the...@sys-concept.com wrote:
>
> On 02/15/2018 04:49 PM, Joshua Colp wrote:
>> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>>
>>
>>
>>>
>>> Thanks again for the hint.
>>> Here is the output from asterisk.
>>>
>>> The call is coming on
On 02/15/2018 04:49 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>
>
>
>>
>> Thanks again for the hint.
>> Here is the output from asterisk.
>>
>> The call is coming on Audocodes gateway from: pstn-
>>
>> But asterisk display:
>> Found peer
On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>
> Thanks again for the hint.
> Here is the output from asterisk.
>
> The call is coming on Audocodes gateway from: pstn-
>
> But asterisk display:
> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
>
> Why not
On 02/15/2018 04:08 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
>> On 02/15/2018 03:44 PM, Joshua Colp wrote:
>>> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
I'm using Audio-codes MP-114 unit and it has two public lines PSTN
On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
> On 02/15/2018 03:44 PM, Joshua Colp wrote:
> > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
> >>
> >> IN audocodes setting I have:
> >>
On 02/15/2018 03:44 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>>
>> IN audocodes setting I have:
>> "EndPoint Phone Number"
>>
>> Channel: 3phone number: pstn-
>>
On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>
> IN audocodes setting I have:
> "EndPoint Phone Number"
>
> Channel: 3phone number: pstn-
> Channel: 4phone number: pstn-9998
>
> When I am
I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
IN audocodes setting I have:
"EndPoint Phone Number"
Channel: 3phone number: pstn-
Channel: 4phone number: pstn-9998
When I am calling " pstn-" the port number "Channel:3" lights up but
asterisk is showing
Hi Olivier,
Yes, it's possible, we have a provisioning like that on production.
I recommend you to follow the Wiki of Gigaset and contact their support in
case of misunderstanding: They help us a lot for the integration.
You only have one XML file to provide, with an example in the wiki, if I
Luca Bertoncello schrieb:
> But if I try to call another VoIP-phone it rings but no voice will be
> transferred...
Got it!
A "little" firewall problem... :(
Regards
Luca Bertoncello
(lucab...@lucabert.de)
--
Tzafrir Cohen schrieb:
> This means that you have configured a dahdi channel in
> /etc/asterisk/chan_dahdi.conf . The default configuration does not
> include one. Do you have any DAHDI device on the system?
I think not...
> If /dev/dahdi/channel itself does not
On Thu, Feb 15, 2018 at 06:55:16PM +0100, Luca Bertoncello wrote:
> Hi again!
>
> I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI
> with Armbian/Debian 9.
>
> First test was to call a test service that say the time. Works!
> Second test was to record my voice and play
Hi again!
I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI
with Armbian/Debian 9.
First test was to call a test service that say the time. Works!
Second test was to record my voice and play it again. Works!
Third test was to call the other VoIP-phone. It does NOT
Zitat von Tzafrir Cohen :
Yes. It is useful if you want to call using a local sound device.
On a Banana PI? ;)
Consider editing /etc/asterisk/modules.conf and disable ('noload =>')
chan_oss.so .
So I did...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
On Thu, Feb 15, 2018 at 08:38:03AM +, Luca Bertoncello wrote:
> Zitat von Tzafrir Cohen :
>
> Hi,
>
> > Off-topic: any reason you don't use chan_alsa?
>
> This was the "Armbian installation", I didn't configured it extra...
>
> > Are you sure you quote the error
Zitat von Tzafrir Cohen :
Hi,
Off-topic: any reason you don't use chan_alsa?
This was the "Armbian installation", I didn't configured it extra...
Are you sure you quote the error message right?
Copy+Paste... ;)
But I searched a little bit and I really don't
Hi,
On Thu, Feb 15, 2018 at 07:45:00AM +, Luca Bertoncello wrote:
> Hi list!
>
> Currently I use Asterisk 1.8.30.0 on an OpenWRT-Switch.
> Now I want to change to Asterisk 13.14.1 on a Banana PI (with Armbian/Debian
> 9).
> Well, I copied the configuration and changed what needed, so
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