Hi!
I have a Asterisk installation to manage my phones at home (provider is
Deutsche Telekom).
It works, but very often the voice is "broken"...
Yesterday during a call it was very difficult to understand what my
partner sayd...
It can NOT be a problem of other downloads/uploads, since in that
Am 13.06.2020 um 13:47 schrieb Michael Keuter:
Hi
> Try "sip show peer " for a phone.
So:
mobile phone:
bpi*CLI> sip show peer 0049177xxx
* Name : 0049177xxx
Description :
Secret :
MD5Secret:
Remote Secret:
Context : default
Record
Am 13.06.2020 um 08:28 schrieb Luca Bertoncello:
> Hi!
>
> I have a Asterisk installation to manage my phones at home (provider is
> Deutsche Telekom).
> It works, but very often the voice is "broken"...
> Yesterday during a call it was very difficult to understand what my
> partner sayd...
>
>
On Saturday 13 June 2020 at 13:36:00, Luca Bertoncello wrote:
> Am 13.06.2020 09:30, schrieb Luca Bertoncello:
>
> Hi again (again)
>
> I noticed right now another strange detail...
> I made a call using my mobile phone (connected to the Asterisk).
What does that mean? You're making a mobile
I'm parsing ` sudo asterisk -rx "core show calls" | grep active | head -c 1
` as an external call from within the Asterisk dialplan then passing it to
agi, but this seems really hacky and ugly.
However, I cannot find any ARI/AGI/AMI function (or global variable I can
get with agi) which shows me
Am 13.06.2020 09:30, schrieb Luca Bertoncello:
Hi again (again)
I noticed right now another strange detail...
I made a call using my mobile phone (connected to the Asterisk). The
quality was top...
Maybe is the problem in a codec used from our phones at homes?
Could someone suggest me how to
> Am 13.06.2020 um 13:36 schrieb Luca Bertoncello :
>
> Am 13.06.2020 09:30, schrieb Luca Bertoncello:
>
> Hi again (again)
>
> I noticed right now another strange detail...
> I made a call using my mobile phone (connected to the Asterisk). The quality
> was top...
> Maybe is the problem in
On Sat, 13 Jun 2020, Jonathan H wrote:
I need to ensure that a MusicOnHold stream is only running when there's
a caller on hold and listening.To do that, I need to rewrite and reload
the moh.conf file when the caller hangs up IF there are no other callers
(ie there's just 1 active call as the
So the call used Alaw as Codec.
> Am 13.06.2020 um 17:23 schrieb Luca Bertoncello :
>
> Am 13.06.2020 um 13:47 schrieb Michael Keuter:
>
> Hi
>
>> Try "sip show peer " for a phone.
>
> So:
>
> mobile phone:
> bpi*CLI> sip show peer 0049177xxx
>
>
>
>
> * Name : 0049177xxx
On Saturday 13 June 2020 at 17:23:14, Luca Bertoncello wrote:
> Am 13.06.2020 um 13:47 schrieb Michael Keuter:
>
> > Try "sip show peer " for a phone.
> bpi*CLI> sip show peer 0049177xxx
> Codecs :
> (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|
>
Am 13.06.2020 um 18:20 schrieb Antony Stone:
Hi
>> bpi*CLI> sip show peer 0049177xxx
>> Codecs :
>> (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|
>> slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|t
>>
On Saturday 13 June 2020 at 18:06:23, Michael Keuter wrote:
> So the call used Alaw as Codec.
...which should be excellent quality.
PS: Michael: thanks for the tips regarding "sip show channels" and "sip show
channel " - I was aware of these for some details, but hadn't realised
they showed
Am 13.06.2020 um 18:06 schrieb Michael Keuter:
> So the call used Alaw as Codec.
Yes, so seems it to be...
It should has the better quality... But the calls done using my mobile
phone in VoIP with the Asterisk have better quality as the calls done
using the normal VoIP-telefon...
I'm really
On Saturday 13 June 2020 at 18:26:53, Luca Bertoncello wrote:
> Am 13.06.2020 um 18:06 schrieb Michael Keuter:
> > So the call used Alaw as Codec.
>
> Yes, so seems it to be...
> It should has the better quality... But the calls done using my mobile
> phone in VoIP with the Asterisk have better
Am 13.06.2020 um 22:09 schrieb Antony Stone:
Hi Antony
> You are *assuming* that it's the codec causing the difference.
Well, I really don't know what I can think, now...
> We don't know that.
>
> Let me get this clear, to make sure I understand (differences emphasised):
>
> 1. You use *a
On Saturday 13 June 2020 at 22:30:28, Luca Bertoncello wrote:
> 1) I have an Android phone, using the integrated Android VoIP-subsystem,
> connected to my Asterisk at home, over LTE or other network *outside my
> home network*.
> I called my mother using this method... The quality was excellent
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