Displaying 16 results from an estimated 16 matches for "fictus".
2016 Sep 08
3
Asterisk 13 and WebRTC
Hello list,
before to lost my time, I'd like know if someone have a WebRTC working
configuration on Asterisk 13.11.0 SIP or PJSIP channel.
Thank you
Regards
2016 Jun 06
4
PJSIP subscribe
Hello,
I'm trying to use presence with PJSIP and I have a "issue".
I created correctly hint priorities like:
exten => 1000,hint,PJSIP/1000
exten => 1001,hint,PJSIP/1001
Extension 1000 can subscribe extension 1001 y vice-versa. The problem is
when the extension 1000 make or receive a call. In the softphone where
the extension is present on buddy list, the extension appear
2016 May 16
2
Asterisk PJSIP Multi-tenant
...send_unsolicited_mwi_notify_to_contact: Unable to create unsolicited
NOTIFY request to endpoint 1001 at sip.domain.com URI
sip:1001 at 95.250.29.3:50673;rinstance=1af959e7c0059fc4
Regards
El 16/05/2016 a las 02:52, George Joseph escribi?:
>
>
> On Sun, May 15, 2016 at 12:00 PM, Annus Fictus <annusfictus at gmail.com
> <mailto:annusfictus at gmail.com>> wrote:
>
> Hello List,
>
> following this thread:
>
> http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains
>
> I tried to configure on the...
2015 Dec 16
2
Help with CDR-Stats
Humm whats is the diferent?
Em 16/12/2015 14:19, "Annus Fictus" <annusfictus at gmail.com> escreveu:
> CDR-STATS is for reporting.
>
> A2Billing is for billing...
>
> Regards
>
> El 16/12/2015 a las 11:15, Vitor Mazuco escribi?:
>
>> Hi everyone!
>>
>> I'm trying to install CDR-Stats (cdr-stats.org), bu...
2015 Dec 02
2
Issues with Twilio number incoming call and context matching
...-external,${EXTEN},1)
[pbx_config]
ip-xxxx*CLI> dialplan show from-external
[ Context 'from-external' created by 'pbx_config' ]
'17775551212' => 1. Log(WARNING,TWILIO)
[pbx_config]
2. Hangup()
[pbx_config]
On Wed, Dec 2, 2015 at 9:23 AM, Annus Fictus <annusfictus at gmail.com> wrote:
> Hello,
>
> try to change:
>
> exten => 17775551212,1,Log(WARNING, TWILIO)
> same => n,Hangup()
>
> with:
>
> exten => +17775551212,1,Log(WARNING, TWILIO)
> same => n,Hangup()
>
>...
2016 Jul 17
3
PJSIP - State of the art
Hello,
I'd like share with you my tests about PJSIP channel with the aim of
improving the functioning of the channel:
* Multi domain support not work correctly:
https://issues.asterisk.org/jira/browse/ASTERISK-26026
* Different context subscribe for each endpoint not possible:
https://issues.asterisk.org/jira/browse/ASTERISK-25471
* BLF don't work correctly on my tests
2016 Jun 13
2
PJSIP does not qualify contacts after starting Asterisk
...ip]
endpoint=realtime,ps_endpoints
aor=realtime,ps_aors
contact=realtime,ps_contacts
[res_pjsip_endpoint_identifier_ip]
identify=realtime,ps_endpoint_id_ips
Cheers, Francisco.
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Annus Fictus
Sent: 13 June 2016 14:11
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] PJSIP does not qualify contacts after starting Asterisk
Hello Francisco,
you have to use:
extensions => odbc,asterisk
only if you wan...
2016 Jan 19
2
Statsd Dialplan Application
Hello,
I'd like to do some tests with the StatsD dialplan application but on
the last version of Asterisk 13 (13.7.0) I can't find this application.
New Features made in this release:
-----------------------------------
* ASTERISK-25419 - Dialplan Application for Integration of StatsD
(Reported by Ashley Sanders)
res_statsd module are correctly compiled y loaded.
Any hint?
2017 Feb 16
2
Soft SIP phones that support TLS - Asterisk version 13.13.1
Hi,
Am 16.02.2017 um 14:19 schrieb Annus Fictus:
> And Microsip using PJSIP SIP stack :)
Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not sure if this really is a sign of good quality.
Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP in other software), but after just f...
2016 Jun 17
2
Agents.conf Device_state
Hello,
I think Device State for Agents don't work correctly
My configuration:
agents.conf
[general]
[agent](!)
autologoff=15
ackcall=no
acceptdtmf=#
wrapuptime=5000
musiconhold=default
recordagentcalls=no
custom_beep=beep
[2000](agent)
fullname=Fulano
[2001](agent)
fullname=Zutano
[2002](agent)
fullname=Mengano
queue.conf (Agents Related)
member => Agent/2000
member =>
2016 May 15
2
Asterisk PJSIP Multi-tenant
Hello List,
following this thread:
http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains
I tried to configure on the pjsip.conf the same endpoint with different
domains like:
[1000 at sip.domain.com]
type=endpoint
[1000 at sip1.domain.com]
type=endpoint
I can register the two 1000 endpoints using different domain but on the
Asterisk console:
2015 Dec 16
2
Help with CDR-Stats
Hi everyone!
I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult.
Is there others optins for billing?
Thanks
2017 Feb 16
2
Soft SIP phones that support TLS - Asterisk version 13.13.1
Microsip (Windows) is free and small.
2.5Mb download, 10Mb RAM usage, does everything I need and configuring
TLS is a doddle.
http://www.microsip.org/
On 16 February 2017 at 13:04, Max Grobecker
<max.grobecker at ml.grobecker.info> wrote:
> Hello,
>
> I'm a big fan of PhonerLite.
> It's more poplar in Germany, but also available in English language.
> This client
2015 Dec 02
4
Issues with Twilio number incoming call and context matching
Hello,
I am running Asterisk 13.6.0 in an AWS instance, and I set it up with
Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the
calls actually "reach" the PBX, but for some reason, they are not caught by
any of my extensions context.
Here's what I observe when I test this from a non-PBX connected E164 number
(a landline), say 555-666-1212. My Twilio number is
2016 Jun 13
2
PJSIP does not qualify contacts after starting Asterisk
Hi all,
(sending this again from the correct address)
I'm running Asterisk 13.8.0 (I need to check if that happens with 13.9.1 too when I have the time to build it) with PJSIP realtime config.
I've defined several aors in the table ps_aors, like this (real url replaced by myurl):
*CLI> pjsip show aor pbx-node-1
Aor: <Aor..............................................>
2016 Sep 12
4
Mysql PJSIP realtime > 13.10?
Has anyone successfully used Mysql realtime PJSIP with Asterisk
13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the
following error now:
Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql:
Realtime table general at ps_contacts: column 'qualify_timeout' cannot be
type 'int(10)' (need char)
[Sep 12 14:42:35] WARNING[24498]: