Hello,
on Asterisk 13.13.1 working correctly
Regards
El 27/02/2017 a las 10:59, Steve Edwards escribió:
Asterisk 13.3.2
I change the allowed codec from ulaw to g729 in sip.conf and enter
'sip reload' on the console, but calls continue to use ulaw until
restart.
Before reload:
lc10*CLI>
And Microsip using PJSIP SIP stack :)
El 16/02/2017 a las 08:15, Jonathan H escribió:
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
Hello,
I'm not totally sure but HEP permit SIP signaling and RTCP data capture
only on PJSIP channels. For chan_sip you have to use captagent.
Regards
El 03/01/2017 a las 10:04, Olivier escribió:
Hello,
On a newly built Asterisk 13.13.1 system, I can't make HEP work with
chan_sip (though
Look at:
https://javiervalencia.net/2015/12/06/asterisk-en-realtime/
(Spanish)
Regards
El 09/11/2016 a las 17:06, Joshua Colp escribió:
On Wed, Nov 9, 2016, at 05:59 PM, Carlos Chavez wrote:
Is there some documentation for all the available sorcery.conf
mappings for realtime?
Hello Carlos,
I'm testing CentOS 7 ODBC packages with PJSIP Realtime without problems.
Maybe you use a different configuration?
Regards
El 12/09/2016 a las 16:01, Carlos Chavez escribió:
On 9/12/16 3:39 PM, George Joseph wrote:
On Mon, Sep 12, 2016 at 2:31 PM, George Joseph
Hello,
is there any reason you don't use ODBC with MySQL?
Regards
El 12/09/2016 a las 15:14, Carlos Chavez escribió:
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
roginvs
https://github.com/DoubangoTelecom/sipml5/pull/238
Dne 08/09/2016 v 23:36 Annus Fictus napsal(a):
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
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
--
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hello,
try to add fromdomain=yourdomain
in your trunk configuration.
regards
El 04/08/2016 a las 21:20, Telium Technical Support escribió:
We are working with an ISP that needs Asterisk to place a FQDN name in
the SIP ‘FROM’ and ‘INVITE’ fields – where Asterisk is currently using
an IP
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:
with templates.
Regards
El 13/07/2016 a las 23:49, Carlos Chavez escribió:
Until Asterisk 11 I could use sip.conf to set defaults for all
phones (language, dtmf, vmexten, etc) and just leave many fields in
the database as NULL. What would be the proper way to do this for
Asterisk 13
hello,
I'm trying to use Asterisk 13.9.1 with Homer SIP Capture Server.
My hep.conf Asterisk configuration is:
[general]
enabled = yes
capture_address=107.170.151.154:9060
;capture_password = foo
capture_id = 2464
SIP Signaling work correctly but no RTCP STATS arrive to Homer Server.
On the
Hello,
I'm using 13.9.1 version
thank you for your answer.
Now working fine.
Regards
El 17/06/2016 a las 20:06, Richard Mudgett escribió:
On Fri, Jun 17, 2016 at 11:50 AM, Annus Fictus <annusfic...@gmail.com
<mailto:annusfic...@gmail.com>> wrote:
Hello,
I think
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
Hello I would like to know if anyone has been able to set up a
workingBLF/Presence configuration with PJSIP channel. If yes, please
share the configuration and Softphones/Phones used. Thank you Regards
--
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-- Bandwidth and
]
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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Annus
Hello Francisco,
you have to use:
extensions => odbc,asterisk
only if you want use dialplan in Realtime
can you share your sorcery.conf file?
Regards
El 13/06/2016 a las 10:21, Francisco Valentin Vinagrero escribió:
Hi all,
(sending this again from the correct address)
I’m running
I have seen the problem is with:
Jitsi, XLite, Bria 3.X
Grandstream Wave work correctly.
I have to test with hardphone.
Regards
El 09/06/2016 a las 01:11, George Joseph escribió:
On Wed, Jun 8, 2016 at 8:48 AM, Annus Fictus <annusfic...@gmail.com
<mailto:annusfic...@gmail.com&g
Hello,
How can I know if is a BUG and report on Asterisk-Jira?
Thank you
Regards
Mensaje reenviado
Asunto: PJSIP subscribe
Fecha: Mon, 6 Jun 2016 19:13:35 +0200
De: Annus Fictus <annusfic...@gmail.com>
Para: Asterisk Users Mailing List - Non-Comm
Hello,
thank you for the answer... how can I see the correct status?
any configuration on asterisk or softphone side?
Regards
El 07/06/2016 a las 16:36, George Joseph escribió:
I can confirm that Bria shows offline but if the client is using the
tuple status instead of the person status then
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.
Hello,
I'm trying to use JABBER_RECEIVE function on my dialplan but the timeout
function don't work.
This is my dialplan:
[google-in]
exten => s,1,NoOp( Call from Gtalk )
same => n,SendText(Hola,Como te llamas?)
same => n,Set(nombre=${JABBER_RECEIVE(google,${CALLERID(name)},30)})
same =>
Done.
ASTERISK-26026
El 16/05/2016 a las 14:40, George Joseph escribió:
On Sun, May 15, 2016 at 10:17 PM, Annus Fictus <annusfic...@gmail.com
<mailto:annusfic...@gmail.com>> wrote:
Hello,
with qualify_frequency=0 I can't receive calls from others endpoints.
O
escribió:
On Sun, May 15, 2016 at 12:00 PM, Annus Fictus <annusfic...@gmail.com
<mailto:annusfic...@gmail.com>> wrote:
Hello List,
following this thread:
http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains
I tried
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:
[1...@sip.domain.com]
type=endpoint
[1...@sip1.domain.com]
type=endpoint
I
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
Thank you for the response.
Regards
El 19/01/2016 a las 11:22, Kevin Harwell escribió:
On Tue, Jan 19, 2016 at 8:46 AM, Annus Fictus <annusfic...@gmail.com
<mailto:annusfic...@gmail.com>> wrote:
Hello,
I'd like to do some tests with the StatsD dialplan
CDR-STATS is for reporting.
A2Billing is for billing...
Regards
El 16/12/2015 a las 11:15, Vitor Mazuco escribió:
Hi everyone!
I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult.
Is there others optins for billing?
Thanks
--
Hello,
try to change:
exten => 17775551212,1,Log(WARNING, TWILIO)
same => n,Hangup()
with:
exten => +17775551212,1,Log(WARNING, TWILIO)
same => n,Hangup()
Regards
--
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-- Bandwidth and Colocation Provided by
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 <annusfic...@gmail.com
<mailto:annusfic...@gmail.com>> wrote:
Hello,
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