Did you have a look at the phone it self already?
Is call forwarding activated or something and can you call the
phone/extension from externally?
I have seen this in the past where an employee enabled call forwarding
on the phone and once at home he or family called the phone which
forwarded the
As we are top posting I will continue this.
Please have a look at:
https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
I hope this answers your questions.
Regards,
Michel.
op 13-08-14 01:34, Rafael Visser schreef:
I am talking about sip on asterisk 11.10.2
op 04-02-14 09:29, sylvain Gotri schreef:
Hi ,
I have asterisk 1.8.5 installed on Centos 6. Now I want to configure
my PBX to work in my network. I see that I can do this with asterisk
files or use database like mysql to do it (realtime)
I want to know what is the best way and what can be
On 09-09-13 23:11, Niccolò Belli wrote:
Hi,
I have a Portech MV-374 GSM Gateway and I'd like to send SMS from a
web page to confirm the subscriptions. How can I achieve it? Is
Asterisk of any use to send SMS with the Portech? I really have no
idea because I know nothing about the whole SMS
Please also have a look at the gateway boxes from berofix
(http://wiki.beronet.com/index.php/Main_Page). I am not affiliated but
have used different products from them over last few yeas and all have
survived and are stable.
Documentation is open and free on their wiki. They provide updates. They
Please also have a look at the gateway boxes from berofix
(http://wiki.beronet.com/index.php/Main_Page). I am not affiliated but
have used different products from them over last few yeas and all have
survived and are stable.
Documentation is open and free on their wiki. They provide updates. They
We use Zabbix as monitoring tool and SNMP to get statistics and other
info from Asterisk.
for this you will have to make sure the snmp module for asterisk gets
compiled and the Asterisk MIB is used.
Regards,
Michel.
On 09-05-13 21:23, motty cruz wrote:
Hello,
i'm looking for suggestions to
Op 08-10-12 15:17, Olivier schreef:
2012/10/8 Michel Verbraak mic...@verbraak.org
mailto:mic...@verbraak.org
Op 08-10-12 09:24, Olivier schreef:
Hi,
I've read this thread in this list history
http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match
Op 08-10-12 09:24, Olivier schreef:
Hi,
I've read this thread in this list history
http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem
http://sourceforge.net/tracker/?func=detailaid=3337581group_id=152230atid=783657
Has anyone been successful when
Op 03-10-12 01:17, Chris Nighswonger schreef:
On Tue, Oct 2, 2012 at 5:30 PM, Chris Bagnall
aster...@lists.minotaur.cc wrote:
On 2/10/12 6:51 pm, Carlos Alvarez wrote:
Your traffic level, number of concurrent calls, etc would help us know
what
sort of carrier you should be talking to.
Op 03-10-12 15:08, Tim Nelson schreef:
- Original Message -
Have a look at your /etc/asterisk/rtp.conf file. In it you specify
the UDP portrange your asterisk will use for RTP traffic. change the
rtpstart and rtpend to your needs and set them open in your FW. Do
not make the range
Op 22-08-12 12:09, Shitian Long schreef:
I am trying to setup TE110P wildcard on a PBX running ubuntu 12.04
server edition. I followed the procedure
from http://docs.digium.com/misc/ADL_quickstart.pdf step by step.
During the process of installing dahdi-linux-complete
I got following
On 30-04-12 11:09, Bharat Lalcheta wrote:
Hiii all,
I am using asterisk 1.8.9.2 and compile all modules related to calendar.
neon version is 0.29.6. OS is ubuntu 11.10.
I configured ical for zimbra, caldav for google mail and ews for
exchange 2010 calendar.
ical and caldav setup working
On 21-04-12 08:19, Olivier CALVANO wrote:
Hi
I have a small problems with incoming call.
I have a peer actually configured for outcall :
sip.conf:
[Trunk-Telco]
type=peer
host=domaineofmysupplier.net
outboundproxy=domaineofmysupplier.net
session-timers=originate
session-expires=7200
Hi,
Use the local channel
Dial(Local/@contextinternallocal/b@contextexternal)
In the internal context you set CALLERID(num) to the internal extension
and then dial the SIP
exten = ,1,Set(CALLERDI(num)=${EXTEN})
same = n,Dial(SIP/${EXTEN})
In the external context do almost
methods? Then you would probably have
covered everyones needs.
Regards
Anders Fudali
From: Michel Verbraak mic...@verbraak.org mailto:mic...@verbraak.org
Date: Sun, 12 Jun 2011 14:59:02 +0200
To: Anders Fudali anders.fud...@jajja.com
mailto:anders.fud...@jajja.com
Subject: Re: [asterisk-users
Asterisk and would
like to have access to any calendar in Exchange please try the patch in
the following review request: https://reviewboard.asterisk.org/r/1152/.
Please reply to the reviewboard if it is working for you or if you
experience problems.
Regards,
Michel Verbraak
Op 27-05-11 17:10, Michelle Dupuis schreef:
I'm looking for recommendations for standalond PRI to SIP converters. (Needs
to be outside the asterisk box - so a PCIe card won't do)
I've used redfone but this project doesn't need the redundancy features...
Thanks!
--
Almost,
If you use Asterisk version 1.6 or higher use
Exten = ExecIf($[${CALLERID(number)} = 400]?Set(CALLERID(num)=)
Or
Exten = ExecIf($[${CALLERID(number)} = 400]?Set(CALLERID(all)=)
Michel Verbraak
**
http://www.intercommit.nl/
On 08-04-11 15:56, Louis Carreiro wrote:
Hey all
We also see the random freeze of asterisk 1.8.3.2. We do use realtime.
I have just applied the patch and will see how our environment holds.
I will report back to the issue mentioned by Ishfaq
Michel Verbraak
*InterCommIT bv* **
On 06-04-11 09:44, Ishfaq Malik wrote:
On Tue, 2011-04-05 at 21
Jeff LaCoursiere schreef:
Working with a new client that has a ton of these phones, and in a new
installation the phone is registered, can place and receive calls with no
issues, but has a locked picture of a phone in the upper right corner.
Any Linksys experts know what this means? I have
trebaum schreef:
I keep getting a red alarm when trying to setup asterisk to use my
TE420B EC. I only have a blank context setup in my extensions.conf as
I haven't started to config that until I can clear this red alarm. I
don't have physical access to the server, so I can't go reseat the
Hello,
First of all I have an Asterisk setup of Asterisk 1.6.0.9 + DAHDI 2.0 +
E1 card with ISDN-15 line (KPN Netherlands).
I have two questions/situations:
A. I would like to be able to interrupt the dial command when I try to
call to a mobile phone and this phone is never answered by a person
Hans Konings schreef:
Hi
I'm having problems getting the TE420 working in HP DL380G5 servers.
The cards don't seem to be detected 100% by the BIOS. With two cards
in the server they are never detected.
things I've tried:
1 Update firmware to latest (P56) for the server
2 change irq
Oguzhan Kayhan schreef:
Oguzhan Kayhan wrote:
I want to change it to E1 instead of T1.
here comes the problem.
If it's anything like the older cards, there is a jumper on the card
that sets it to T1/E1
Doug
Yes,
I just noticed the jumper on the card.
Thanks a lot.
25 matches
Mail list logo
