Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

2017-06-27 Thread Teijo

Hello,

Yes. When I today understood to set rtcp_mux=yes, at least Chrome (60.0 
beta) worked (quickly tested) as expected.


I'm sure that some day dtls_rekey can be set to the other value than 0 
as well with Chrome.


Best regards,

Teijo

10.4.2017, 16.57, Matt Fredrickson kirjoitti:


On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkins  wrote:



On Fri, Apr 7, 2017 at 9:44 PM, Teijo  wrote:


Hello,

I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
problem until now which remained was that if dtls_rekey was set to the
value other than 0, call hanged up when using chrome after the time where
dtls_rekey was set.

I suppose that "bad media description" shown in Chrome's window which
causes call to fail, has appeared with Chromes newer versions (currently 58
beta installed) or with Asterisk 13.15.0. Audio codec I'm using is Opus.

Has somebody else encountered this problem, or more better resolved it?

Best regards,

Teijo

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Hi Teijo

Take a read of https://nimblea.pe/monkey-business/2017/01/19/webrtc-
asterisk-and-chrome-57/ :)



13.15.0 should address rtcp-mux issues.

If there are still issues outstanding, it might be worth reporting a bug on
issues.asterisk.org.

Best wishes :-)



Tämä osa viestin runkoa ladataan pyydettäessä.



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Re: [asterisk-users] Autodialer - call simultaneously to both ends

2017-06-27 Thread Harel
Thank you Jean, Antony and J Montoya or A J Stiles for replying.
Jean: My Asterisk is version 1.8 therefore I don't think it supports ARI
Antony: In general you are right about calling local channel first however if I 
issue the call setup simultaneously in 99.9% of the cases the local agent will 
answer way before the remote side will and I'm willing to sacrifice this 0.1% 
for this purpose. The reason for this is that the local agents are on a queue. 
Would you know how to Originate() or callfile to a queue? AMI is an option I'll 
look at.
JM or AJS: Yes, this is how it is working for me (including the move vs copy) 
so my problem is not there.

My main objective is to initiate a call as soon as an agent becomes available 
(also using predictive algorithms) and I don't want to waste time on calling 
one side and only when answered calling the other.

Harel
-- Original Message 
---

Message: 1
Date: Mon, 26 Jun 2017 15:06:10 +0300
From: "Harel" 
To: 
Subject: [asterisk-users] Autodialer - call simultaneously to both
ends
Message-ID: <012b01d2ee74$9a52cb60$cef86220$@mayorcom.com>
Content-Type: text/plain;   charset="utf-8"

Hello List,
I'm working on an autodialer project.
At the moment I use the Originate application then I "throw" it to an extension 
where I Dial() the other party and then both legs are bridged. The problem is 
that the Dial() will only run after the Originate finish its bit and I have 
lots of wasted time or even worse, the remote party hanging the call because 
instead of a human speaking to him the call setup to the 2nd leg only starts 
when remote answers.
Is there a way to start calling both parties at the same time and bridge them 
when one answers (which will then hear the ringback tone until 2nd party 
answers)?
Thank you
Harel




--
From: Jean Aunis 

Hello,

You can certainly do this with the ARI interface but you will have to write 
some code.

You could for example originate two channels, send them to the same Stasis 
application, create a bridge and place both channels in it.

Best regards

Jean Aunis

--
From: Antony Stone 

Sounds like you're dialling the legs the wrong way round.

You should dial the extension of the person who wants the autodial function 
first (ie: the person who knows about this system).

They answer their phone (which should be quick, because they're expecting it to 
ring after they've initiated the autodial), and they then wait for the remote 
party (who doesn't know there's an autodialler involved) to answer.

Dialling both numbers simultaneously always runs the risk that the remote party 
(who doesn't know about the autodialler) will answer the call first, so unless 
you have some recorded announcement "please wait while we connect your call" 
(which if I heard it would make me hang up immediately, because I'd know it was 
some automated dialler, probably a cold-calling sales organisation), they 
answer the phone, hear ringing, think "what the hell?" or even "oh, one of them 
again" and hang up.

Always start from the "local" end - ie: the person who knows about the auto- 
dialler.



Antony.

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Atheism is a non-prophet-making organisation.

   Please reply to the list;
 please *don't* CC me.



--
From: J Montoya or A J Stiles 


Our auto-dialler works as follows;

* Agent clicks number on screen in their web browser
* Agent's phone rings
* Agent picks up phone
* Far end party's phone rings
* Far end party answers
* Agent and far end party are bridged.

and is implemented using the truly ancient technology of callfiles.


All you need then is a Perl or PHP script, which accepts the destination 
number as a query parameter.  Your script then needs to identify the 
workstation by means of its IP address and determine the number of the nearest 
phone  (this does require proper configuration of DHCP server, but is worth 
it),  then write out a callfile.


Note:  There exists a race condition in Asterisk  (at least, when using the 
common Linux file systems, which update a folder's directory as soon as the 
*first* block of a file is written)  which means that if a callfile exceeds one 
block, Asterisk could end up reading only the first block and ignoring the 
rest.  If there is any danger that a callfile could exceed one block on your 
filesystem, you must write the callfile to a different folder, and then use the 
`mv` command to move it to /var/spool/asterisk/outgoing/ .  This sidesteps the 
race condition due to the behaviour of the mv command.  When moving *within* a 
filesystem, the whole file was already on the disk anyway when the directory is 
updated; when moving from one filesystem to another, it does not update the 
directory of the destination folder until the *last* block 

Re: [asterisk-users] pjsip configuration realtime+static

2017-06-27 Thread Joshua Colp
On Tue, Jun 27, 2017, at 05:20 AM, marek cervenka wrote:
> hi,
> 
> i have mix of realtime and static configuration of pjsip
> 
> https://pastebin.com/YVFwVsMD
> 



> 
> but the section [global] is not honored

You don't have a "type=global" line in the global section so it's not
being found.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] pjsip configuration realtime+static

2017-06-27 Thread marek cervenka

hi,

i have mix of realtime and static configuration of pjsip

https://pastebin.com/YVFwVsMD

pjsip.conf
[global]
endpoint_identifier_order=username,ip,anonymous
user_agent=ipbx
...
transport definition
extconfig.conf
[settings]
ps_endpoints => odbc,configDb
ps_auths => odbc,configDb
ps_domain_aliases => odbc,configDb
ps_aors => odbc,configDb
sorcery.conf
[res_pjsip]
endpoint/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes
endpoint=realtime,ps_endpoints
auth/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes
auth=realtime,ps_auths
aor/cache=memory_cache,expire_on_reload=yes,full_backend_cache=yes
aor=realtime,ps_aors
domain_alias=realtime,ps_domain_aliases

but the section [global] is not honored

CLI> pjsip show settings

Global Settings:

 ParameterName   : ParameterValue
 ===
 contact_expiration_check_interval   : 30
 debug   : no
 default_from_user   : asterisk
 default_outbound_endpoint   : default_outbound_endpoint
 default_realm   : asterisk
 default_voicemail_extension :
 disable_multi_domain: false
 endpoint_identifier_order   : ip,username,anonymous

--snip--

 user_agent  : Asterisk PBX 13.16.0

any ideas?

Marek


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