Hi,
In my case I used them both. It depends on the features I am going to activate.
There are some departments that use only a standard PABX-like, use 8 Ports FXO
Gateway, 48 Ports FXS Gateway and all they need is to be able to call and be
called. I used Trixbox for this application. I have
Hi Lito,
It depends on how you asked your telco provider to configure your 3 direct
lines. We called it trunking the 3 lines with pilot number. Telco can figure
it the way we configure our Asterisk Followme. (seized all, random, sequencial).
Regards,
Angel
Lito Lampitoc [EMAIL PROTECTED]
Hi Hind,
Sorry, I haven't work for it yet. I still left on the number of endpoints
assigned. Probably I will concentrate on it once I finish with my gnuDialer
Project.
I'll keep you informed.
Angel
hind habaoui [EMAIL PROTECTED] wrote: hi angel.
it is about the CallerId, i have the same
Hi Yehavi,
Yes, this can be done. We are currently using this features. The Secretaries
making the calls to who ever her Boss wants to join the conference she then
just transfer the calls into the conference room. You can even annouce the name
of the newly arrived calls in the conference.
Hi Artur,
Just follow the information Moises had recommended you and for sure this will
work. The default configurations that was exampled in the document is just
fined and suited with Nortel Meridian. Just be sure that your Nortel MFC Card
is installed and working in good condition with up to
Hi Giorgio,
I guess it will be more benefitial to all old version users to read some
information regarding new version like * 1.4. In this way, they will be
encourage, or probably have an idea whether to upgrade or not based on all the
concerns that was posted. I for one still using 1.2.13
- Automatic Dialing - Define a station number located on Asterisk /
Trixbox (ie 101) for all ports
- Caller ID - Allowed .. turn off if you want to Identify the line
they came in on.
- Detect Caller ID from Tel - Enable
Thanks,
Steve Totaro
From: Angel Heart
Reply-To: Asterisk
solve this issue?
I have the same problem.
Thanks,
José
El jue, 01-02-2007 a las 01:15 -0800, Angel Heart escribió:
Hi,
I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming
outgoing calls. However, I noticed that the caller ID of the caller
coming from the FXO displays
Hi Allen All,
I had posted this kind of problem 2 weeks ago but seems nobody from here
encountered yet. So I haven't received any reaction as of the moment.
The problem with AudioCodes' FXO is that I cannot make it work without defining
endpoints number. Once a number is defined, this number
Hi,
cud any one help me figuring out the problem... When the agent in a queue is
engaged, it cannot accept anymore calls, below is the script;
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/2063||tr) in new stack
-- Called 2063
=custom/client-in-queue
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=
maxlen=0
leavewhenempty=no
joinempty=Yes
context=ivr-6
announce-holdtime=yes
announce-frequency=30
Kindest regards.
Angel
Ex Vitorino [EMAIL PROTECTED] wrote: On 2/15/07, Angel Heart wrote:
cud any one
Hi,
What Network Switch you are using? I do traffic/bandwidth shapping on the edge
switch where the port the voice installed, you can configure each port to
128Kbps or just plain Ethernet Port. So the link between bldg. will always be
10Mb/s, who ever uses it whether data or voice and enable
Hi,
I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming
outgoing calls. However, I noticed that the caller ID of the caller coming from
the FXO displays its endpoints assigned number and not the actual caller's ID
coming from PSTN.
Hope someone is using the same scenario
theory (out of the book).
But bottom line is, it works. Magic !
Angel.
Facundo Ameal [EMAIL PROTECTED] wrote: Thanks for the response, I 've already
matched codecs. I have no
problems with that. Do rxgain and txgain have something to do with R2
protocol errors?
Regards.
On 1/28/07, Angel Heart
Hi Facundo,
Were you able to match your phone's codec with the asterisk codec? Try to
check and set them with the same codec. Also, try to adjust the rxgain txgain.
Regards,
Angel
Facundo Ameal [EMAIL PROTECTED] wrote:
Moises,
I 've stated testing by raising all timers a
Hi,
I am using these model from HP ProCurve
http://www.hp.com/rnd/products/switches/switch2600series/features.htm?jumpid=reg_R1002_USEN
http://www.hp.com/rnd/products/switches/ProCurve_Switch_3500yl-5400zl_Series/features.htm?jumpid=reg_R1002_USEN
Regards,
Angel
I don't think if somebody making upgrades for the unicall in accordance to the
latest version of Asterisk. The latest patches of unicall and MFCR2 that I saw
is still for Asterisk ver. 1.2.0. Haven't see any patches for latest version
yet.
This what making me afraid of going to upgrade our
Hi Paul Eric,
Thank you for you information and quick response. I had enabled Monitoring in
every SIP phone already. Made some Playback see below truncated config;
exten = s,21,Dial(${pre_num:4}${the_num}${post_num},120,${TRUNK_OPTIONS})
exten = s,22,Goto(s-${DIALSTATUS},1)
exten =
Hi,
How could I possibly inform incoming callers that the number they'd dialed is
monitored and recorded.
I wanted that when a call-in or call-out is made, a playback will be played to
inform caller callee that thier line is monitored prior to start conversation.
Thanks.
Angel
:09AM -0800, Angel Heart wrote:
Hi,
In what Asterisk file can I edit for me to temporarily unload such
modules. But later I woudl still be able to load them.
Works fine as long as the module is not in use.
asterisk -rx 'unload app_test.so'
Later on:
asterisk -rx 'load app_test.so
Hi,
In what Asterisk file can I edit for me to temporarily unload such modules. But
later I woudl still be able to load them.
Thanks
Angel
Yahoo! Music Unlimited
Access over 1 million songs.
Hi,
You may want to visit www.procurve.com and look for thier training section
there are lots of training materials that can be downloaded. Prices are also
posted in this website.
Actually, all networking manufacturers has thier training docs posted in their
websites.
www.3com.com
Hi,
I am using Procurve Switches by HP for PoE.
http://www.hp.com/rnd/products/switches/ProCurve_Switch_3500yl-5400zl_Series/overview.htm?jumpid=reg_R1002_USEN
Aside from being a LIFETIME WARRANTY, I found them very easy to configure and
install.
Regards,
Angel
- Original Message
Hi guys,
We are using AudioCodes, but still looking an alternative a cheaper one for our
expansion. We are currenly running 4x24-ports FXS VoIP Gateway with 2 Asterisk
Server each server has Dual-Port Card interfaced with E1 PRI ISDN to PSTN and
E1 MFC/R2 to PABX.
Our E1 ISDNs to PSTN are
I am in the middle of installing chan_unicall.c and channels_Makefile.patch in
my channels directory as instructed at the bottom of this doc/site:
http://soft-switch.org/unicall/installing-mfcr2.html. I am at lost 'coz I don't
know where is my channel directory is 'til someone told me that it
Hi Moises,
1. On what directory/folder should I copy the chan_unicall.c, channels_makefile.patch?into wherever you had put the source code of your asteriskinstallation. There, you must have a folder named "channels", whereyou will see several files named chan_sip, chan_zap and in
Hi,
Anyone there could figure me out on how to install my unicall. I followed the instruction belowin the statedsite at; http://soft-switch.org/unicall/installing-mfcr2.html.
Questions:
1. On what directory/folder should I copy the chan_unicall.c, channels_makefile.patch?
2. On what
Hi,
Could anyone knows what this error codes means;
-- Got SIP response 481 "Call/Transaction Does Not Exist" back from SIP Gateway IP AddressThanks
Angel___
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asterisk-users mailing list
To
, 2006 5:43:12 PMSubject: Re: [asterisk-users] Unicall Installation
On Mon, Oct 23, 2006 at 02:11:22AM -0700, Angel Heart wrote: Hi, Thank you for your comment; Below was the result of./configure checking how to run the C++ preprocessor... /lib/cpp configure: error: C++ preprocessor "/li
Hi,
Could anyone knows whatwent wrong with theerror below result of installation of libsupertone.
[EMAIL PROTECTED] latest]# tar xvf
anks again.
- Original Message From: Hadley Rich [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Monday, October 23, 2006 5:01:16 PMSubject: Re: [asterisk-users] Unicall Installation
On Monday 23 October 2006 21:45, Angel Heart wrote:
Hi Guys,Anyone can tell where can I look all your previous post, I am wondering what could my zapata.conf be if I wanted to use two(2) different Trunk Protocol (ISDN R2) in a single Dual Port Digium Card. Sorry, I'm a new user in this forum and new asterisk user as well. Hope somebody
Hello Guys,Same thing with RR, we are currently intalling Asterisk with Digium TE210P Dual Card. We wanted to interconnect its one(1) E1 Port to a Nortel Meridian PABX and we have this kind of status at PABX side; (see below)DTI2 LOOP 18 - ENBL REF CLK: DSBLSERVICE RESTORE: YES ALARM
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