Re: [asterisk-users] Wanted: WebRTC tutorial

[Bruce Ferrell]

On 04/24/2018 09:08 AM, Matt Fredrickson wrote:

On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell  wrote:

A while back (last year maybe?), there was a Digium blog post on setting up
WebRTC.

I was never able to get that working.

I was working with Asterisk 15 on a RHEL derived distro and had no idea of
where to go to shoot the failure.

Has anyone got a tutorial with trouble shooting?

Great question!  I'm assuming you're talking about the SFU blog post -
http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/
?

I'd be curious as to what difficulties you ran into.  We actually need
to try to consolidate the information in that post with the webrtc
setup page on the wiki -
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support

You might try those two pages if you haven't yet.  If you have
already, perhaps posting your specific challenges that you encountered
here might be helpful.

Thanks!


OK, I've gone back and refreshed myself;  When I try to access cybermega in 
/var/lib/asterisk/static-http at port 8088 the asterisk debug shows:

[Apr 24 20:34:48] DEBUG[17170] http.c: HTTP opening session.  Top level
[Apr 24 20:34:48] DEBUG[17170] http.c: HTTP Request URI is /cyber/index.html
[Apr 24 20:34:48] DEBUG[17170] http.c: Requested URI [/cyber/index.html] has no 
handler
[Apr 24 20:34:48] DEBUG[17170] http.c: HTTP keeping session open. 
status_code:404

When I serve it from apache, the web ui appears, but never connects.

Using the firefox dev tools/console I see firefox can't establish a connection the 
server at wss://:8089/ws

The asterisk debug log shows:

[Apr 24 20:39:21] DEBUG[19041] http.c: HTTP opening session.  Top level
[Apr 24 20:39:21] DEBUG[19041] http.c: HTTP Request URI is /ws
[Apr 24 20:39:21] DEBUG[19041] http.c: Requested URI [/ws] has no handler
[Apr 24 20:39:21] DEBUG[19041] http.c: HTTP keeping session open. 
status_code:404

Suggestions?




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Re: [asterisk-users] Wanted: WebRTC tutorial

[Bruce Ferrell]

On 04/24/2018 09:08 AM, Matt Fredrickson wrote:

On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell  wrote:

A while back (last year maybe?), there was a Digium blog post on setting up
WebRTC.

I was never able to get that working.

I was working with Asterisk 15 on a RHEL derived distro and had no idea of
where to go to shoot the failure.

Has anyone got a tutorial with trouble shooting?

Great question!  I'm assuming you're talking about the SFU blog post -
http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/
?

I'd be curious as to what difficulties you ran into.  We actually need
to try to consolidate the information in that post with the webrtc
setup page on the wiki -
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support

You might try those two pages if you haven't yet.  If you have
already, perhaps posting your specific challenges that you encountered
here might be helpful.

Thanks!


Matt,

That is indeed the post.  I could get as far as the second web page screenshot 
and nothing past that login/connect screen... and no meaningful logs on the 
asterisk instance.

I tried serving the client via the Apache instance on the server (2.2 and 2.4) 
and from the asterisk built in http(s) server.

I'm basically a hobbyist and occasional contractor.   The paying job beckoned, 
so...

I'd REALLY like to get it working.  And for the record, I REALLY HATE pjsip.

I've been twiddling Asterisk (and other VOIP systems) since 2002; Linux since '93 and telecom since 1980.  The config is so opaque, poorly documented and error prone I, to this 
day, use the legacy sip config wherever I can.  No one has ever been able to show me an advantage for it and it doesn't seem to use realtime configuration (even more of a 
drawback).  I much prefer realtime for my configuration on Asterisk;  Having configuration picked up from a DB is far preferable to reloading flat files.




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Re: [asterisk-users] Wanted: WebRTC tutorial

[Matt Fredrickson]

On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell  wrote:
> A while back (last year maybe?), there was a Digium blog post on setting up
> WebRTC.
>
> I was never able to get that working.
>
> I was working with Asterisk 15 on a RHEL derived distro and had no idea of
> where to go to shoot the failure.
>
> Has anyone got a tutorial with trouble shooting?

Great question!  I'm assuming you're talking about the SFU blog post -
http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/
?

I'd be curious as to what difficulties you ran into.  We actually need
to try to consolidate the information in that post with the webrtc
setup page on the wiki -
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support

You might try those two pages if you haven't yet.  If you have
already, perhaps posting your specific challenges that you encountered
here might be helpful.

Thanks!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] Wanted: WebRTC tutorial

[Bruce Ferrell]

A while back (last year maybe?), there was a Digium blog post on setting up 
WebRTC.

I was never able to get that working.

I was working with Asterisk 15 on a RHEL derived distro and had no idea of 
where to go to shoot the failure.

Has anyone got a tutorial with trouble shooting?




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[asterisk-users] Asterisk Community Services Outages

[Matt Fredrickson]

Dear Asterisk Community,

For the past 24 hours or so, Digium’s upstream provider has had a few
outages that have affected Asterisk community services, including
Asterisk.org, the mailing lists, and potentially other services.  We
apologize for any inconvenience that it has caused.  Hopefully things
are back up and running, but please let me know if you see anything
that’s still down.

Thanks so much for your patience.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] Alias for country in indications.conf

[Patrick Wakano]

Just did Tzafrir suggestion and it worked like a charm!
Thanks very much!
Cheers,
Patrick Wakano


Virus-free.
www.avg.com

<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

On 23 April 2018 at 23:45, Patrick Wakano  wrote:

> That's quite interesting Tzafrir!
> I will give it a try!
> Thank you very much for the idea!
> Cheers,
> Patrick Wakano
>
>
> On 23 April 2018 at 23:42, Tzafrir Cohen  wrote:
>
>> Also,
>>
>> On Mon, Apr 23, 2018 at 04:08:58PM +1000, Patrick Wakano wrote:
>> > Hello list,
>> > Hope you all doing fine!
>> > I've tried to use the 'alias' directive in the indications.conf file but
>> > apparently it doesn't work
>> > It looks like maybe this feature was removed, because old sample for the
>> > indications.conf file have example using the alias parameter, but newer
>> > samples don't have it anymore also I couldn't find any ticket saying
>> > this parameter was deprecated Anyway when trying to use it, it
>> doesn't
>> > work. Anyone aware of some change related to this?
>> > I am using Asterisk 13.6.0 and have this in indications.conf:
>> > [uk]
>> > *alias = gb*
>>
>> Given that aliases don't work, you can alternatively use:
>>
>> ;;;
>> [uk]
>> description=
>> ringcadence=
>> ...
>>
>> [gb](uk)
>> ;;;
>>
>> --
>>Tzafrir Cohen
>> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
>> http://www.xorcom.com
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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