Re: [asterisk-users] Asterisk / pjsip: RTP - alaw - a=silenceSupp:off

[Joshua C. Colp]

On Wed, Jun 30, 2021 at 1:36 PM Michael Maier  wrote:

>
> Hello!
>
> Short question: Is it possible to set
>
> a=silenceSupp:off
>
> in the SDP for alaw / ulaw for fax calls?
>

No.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
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[asterisk-users] Asterisk / pjsip: RTP - alaw - a=silenceSupp:off

[Michael Maier]

Hello!

Short question: Is it possible to set

a=silenceSupp:off

in the SDP for alaw / ulaw for fax calls?


Thanks
Michael

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Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

[Joshua C. Colp]

On Wed, Jun 30, 2021 at 3:28 PM Jonas Kellens 
wrote:

> Hello
>
>
> I see the following event from the Asterisk Manager :
>
> 2021-06-30 11:20:55
> Array
> (
> [0] => Event: PeerStatus
> [1] => Privilege: system,all
> [2] => SystemName: tstv7
> [3] => ChannelType: SIP
> [4] => Peer: SIP/testacc7700921
> [5] => PeerStatus: Unregistered
> [6] => Cause: Expired
> )
>
>
The cause is in this message, the registration expired. A re-registration
did not occur before the registration expiration so it expired and was
unregistered.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
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[asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

[Jonas Kellens]

Hello


I see the following event from the Asterisk Manager :

2021-06-30 11:20:55
Array
(
    [0] => Event: PeerStatus
    [1] => Privilege: system,all
    [2] => SystemName: tstv7
    [3] => ChannelType: SIP
    [4] => Peer: SIP/testacc7700921
    [5] => PeerStatus: Unregistered
    [6] => Cause: Expired
)

But I see no SIP REGISTER with SIP-header Expires:0 (so an UNregister if 
you like) in my SIP debug.



What I do see is a SIP OPTION, following a SIP 200 OK (so this is the 
qualify frequenty) at 11:20:45



[Jun 30 11:20:45] VERBOSE[1581] chan_sip.c: Reliably Transmitting (NAT) 
to my.lo.cal.ip:55014:

OPTIONS sip:testacc7700921@192.168.1.9:5060 SIP/2.0
Via: SIP/2.0/UDP my.aste.risk.ip:5060;branch=z9hG4bK357689ec;rport
Max-Forwards: 70
From: "asterisk" ;tag=as0da9cbe1
To: 
Contact: 
Call-ID: 0b8c7f5745051ae73193231f5a487...@my.aste.risk.ip:5060
CSeq: 102 OPTIONS
User-Agent: TSTv7
Date: Wed, 30 Jun 2021 09:20:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
Content-Length: 0


---
[Jun 30 11:20:45] VERBOSE[1581] chan_sip.c:
<--- SIP read from UDP:my.lo.cal.ip:55014 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP my.aste.risk.ip:5060;branch=z9hG4bK357689ec;rport=5060
From: "asterisk" ;tag=as0da9cbe1
To: ;tag=2924269434
Call-ID: 0b8c7f5745051ae73193231f5a487...@my.aste.risk.ip:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T46G 28.83.0.120
Content-Length: 0



In sip.conf I have the following config concerning SIP registration expiry :

maxexpiry=3600
;minexpiry=60
;defaultexpiry=120
;submaxexpiry=3600
;subminexpiry=60
qualifyfreq=120


So my question is : what causes the Asterisk Manager to report a 
"PeerStatus: Unregistered" if I find no such data in my SIP debug 
information ??




Kind regards.


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Re: [asterisk-users] ControlPlayBack

[Sean Bright]

On 6/30/2021 9:50 AM, Dovid Bender wrote:
> Yes that works. It's an "ugly hack". Would this be classified as a bug
> or feature?

It's an existing bug:

https://issues.asterisk.org/jira/browse/ASTERISK-27871

Kind regards,
Sean


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[asterisk-users] Combine audio and video from two different sources

[Ryan Press]

I'm trying to combine audio and video from two different sources.  The
application is a door intercom system with a separate RTSP video camera.
I'd like the door intercom to be able to call an extension and that
receiving extension will show the video and have two-way audio.  Also it
would be nice to be able to call the door intercom as the answering device
and show the video as well.

I have this working when the door intercom is the answering device.  I've
found some source code which bridges an RTSP stream to an Asterisk
channel.  Using this I have a working solution with this dialplan:

[from-internal-custom] ; Doorbell video bridge
exten => doorbell_rtsp,1,Answer() same => n,RTSP-SIP(rtsp://
admin:12345@192.168.24.53:554/live/sub,0,asterisk,5060)
; Doorbell combined video/audio incoming
exten => 762,1,Answer() same =>
n,Page(PJSIP/805/doorbell_rtsp@from-internal-custom,qd)

The problem comes when I try to get the door intercom to call an
extension.  Because the calling device (door intercom) does not include
video, the video capability is not added to the ConfBridge and therefore
when I bridge in the RTSP-SIP channel it does not connect.  When I tried to
use Originate it had the same problem, no video codec was offered.

Is there some way I can create a ConfBridge and force a video codec?  Or
use Originate with a video codec?

I see that Asterisk has included Streams for a while now.  Maybe this is
the best way forward but I'm not sure this is something I can easily
configure without writing a bunch of new code.

Thanks,
Ryan
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Re: [asterisk-users] ControlPlayBack

[Dovid Bender]

Sean,

Yes that works. It's an "ugly hack". Would this be classified as a bug or
feature?


On Wed, Jun 30, 2021 at 9:30 AM Sean Bright  wrote:

> On 6/30/2021 8:55 AM, Dovid Bender wrote:
> > [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:779
> > ast_openstream_full: File http://localhost/test.gsm?foo=bar
> >  does not exist in any format
> > [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:1252
> > ast_streamfile: Unable to open http://localhost/test.gsm?foo=bar
> >  (format (ulaw)): No such file or
> > directory
>
> Asterisk does not parse the URLs currently. Maybe something like this
> would work (untested)?
>
> ControlPlayback(http://localhost/test.gsm?foo=bar&_unused=test.gsm
>
> Kind regards,
> Sean
>
>
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>
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] ControlPlayBack

[Sean Bright]

On 6/30/2021 8:55 AM, Dovid Bender wrote:
> [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:779
> ast_openstream_full: File http://localhost/test.gsm?foo=bar
>  does not exist in any format
> [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:1252
> ast_streamfile: Unable to open http://localhost/test.gsm?foo=bar
>  (format (ulaw)): No such file or
> directory

Asterisk does not parse the URLs currently. Maybe something like this
would work (untested)?

ControlPlayback(http://localhost/test.gsm?foo=bar&_unused=test.gsm

Kind regards,
Sean


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[asterisk-users] ControlPlayBack

[Dovid Bender]

Hi,

I am trying to use ControlPlayBack but pass along some values in the GET
request. For instance if I try
ControlPlayBack(http://localhost/test.gsm?foo=bar)
I get an error in Asterisk
[2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:779
ast_openstream_full: File http://localhost/test.gsm?foo=bar does not exist
in any format
[2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:1252
ast_streamfile: Unable to open http://localhost/test.gsm?foo=bar (format
(ulaw)): No such file or directory

The error of the file not existing in the correct format tells me that
Asterisk was able to get the file however it could not play it, since it
didn't know the format. Looking in the /tmp/ dir I see a file called
"bucket-rjS8GT.gsm?foo=bar". I assume Asterisk looks to play it but can't
since it can't figure out the format since it thinks the file ends in
?foo=bar which is not a valid sound file. Is this a bug in asterisk's
implementation of curl so it saves all files only up till the ? or will
that break things for others? For now my dirty hack is to add to the url
=somefile.gsm so the request looks like this:
ControlPlayBack(http://localhost/test.gsm?foo=bar=whocares.gsm)
which saves the file as bucket-2S6Bpl.gsm and that fixes it for me. It's an
ugly hack that I don't like but it works.

Any thoughts?
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