On 13 Feb 2014, at 09:55, Aldo Bergamini aabe...@gmail.com wrote:
Hi,
I did compile the latest DAHDI and LibPRI, with no success… So I thought
about updating the Asterisk package to the last known 1.6.2 release.
Now it's crashing at some different point.
This is the the strace
On 12 Feb 2014, at 23:22, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Try:
# standard asterisk command-line. No verbosity
strace -eopen asterisk -U asterisk -c
See which module was the one last loaded.
--
Tzafrir Cohen
Hi,
I did compile the latest DAHDI and
Hi List,
it feels silly, but here I am.
My Asterisk box is useless, after running a long delayed yum update (Centos
box).
*
A few details on the box:
cat /etc/redhat-release
CentOS release 5.10 (Final)
arch
i686
uname -a
Linux hermes 2.6.18-371.4.1.el5 #1 SMP
On 28 Aug 2013, at 09:50, Grant Bagdasarian g...@cm.nl wrote:
Hi Grant!
I do not know of a way to have multiple 'h' extensions in the same context.
But you can easily make an appropriate context for your custom need!
exten = _X.,1,Playback(invalid)
exten = _X.,n,Hangup
exten =
On 15 Nov 2012, at 15:44, Michael wrote:
Hi Aldo,
Thank you very much for answering my question.
Can you kindly elaborate on how to do the following or at least where to read
about the way to do it?
Hi Michael,
sure...
I am sending you -by direct mail- a diagram that tries to
On 15 Nov 2012, at 14:21, Michael wrote:
Hello,
Does anyone know if it's possible to setup the following scenario?
1. A specific ext(let's say 111) is on active call with an external number
via SIP (let's say 22334455).
2. Via a web GUI, send to asterisk another phone number (22556677)
On 8 Nov 2012, at 10:46, martin f krafft madd...@madduck.net wrote:
For a 3 way conference, all those days phones are able to do this.
Yeah, except I want Asterisk to handle that, not my phone (which
might lose reception or run out of battery etc.).
Martin,
I understand that having the
On 18 Oct 2012, at 17:19, Michelle Dupuis wrote:
I need to do this from the AMI (not the CLI)...I don't *think* a comparable
command exists from the AMI.
As well, I don't want to poll the system for calls so I'm hoping to trap a
call bridged,unbridged type event.
Michelle,
if you
On 18 Oct 2012, at 17:58, Jerry Geis wrote:
I was wanting to call ChanIsAvail from AMI.
I logged in and issues command,
Action: Command
Command: ChanIsAvail DAHDI/1
my response was this:
event_list=0 ret=158 Response: Follows[CR ][LF ]Privilege: Command[CR ][LF
]No such command
On 28 Sep 2012, at 13:27, Leif Madsen leif.mad...@asteriskdocs.org wrote:
Generally the preferred method when you're doing this programatically anyways
is to use an external script through the Asterisk Manager Interface to
generate your calls.
Luckily, Russell Bryant has recently create
On 28 Sep 2012, at 14:24, Leif Madsen leif.mad...@asteriskdocs.org wrote:
That's good! I'd hate to be working on something no one wanted :)
;-)))
Oh heck ya. You can start up an Asterisk instance and just start doing things
with it via your programs. That's the immense power of AMI; it's
On 21 Aug 2012, at 19:32, Ruben Rögels ruben.roeg...@jumping-frog.org wrote:
Hello,
no problem at all, I think this is the tricky part.
A smtp dialogue between your email client and a smtp server normally looks
like this:
user@box:~? netcat mx1.example.com
220 postfix ESMTP
On 12 Apr 2010, at 22:14, asterisk-users-requ...@lists.digium.com wrote:
There's system clock, and hardware clock.
Whatever you get for the localtime when you do 'date' command is what
you're going to get for logs from asterisk.
It seems somewhere you have your system set to run in GMT,
Hi all,
I have noticed something I can't solve regarding Asterisk (latest
1.6.0.x).
My server is set at the GMT+2 timezone. The clock is ok (I can get the
correct time at the terminal). But today I got a call at a time where
Asterisk should have gone 'off business hours'.
All log times
Hi,
I am seeking help with the installation of a Soundpoint 650 desk phone.
Although I have some experience (and a good one! no single issue so
far, besides the problem I am trying to solve...) installing a few SP
320/330 units, I am having several issues with my first SP 650.
Polycom SP
Hello List,
it is a very long time since I wrote here It has been still in
Zaptel times
Today I am run into a related problem: I can't get a DAHDI setup to
work 100%. I am configuring an Astribank XR00013 (BRI, two ISDN ports).
At some degree the installation (latest DAHDI drivers,
[EMAIL PROTECTED] is believed to have said:
We are currently writing a reception console for Asterisk - if anyone is
interested in beta testing it, feel free to ask.
Paul Hales
Does it run on *nix (Linux/MacOSX)?
Is there a place we can see some information without cluttering the
[EMAIL PROTECTED] is believed to have said:
I'd expect it to be in Falcom's best interest to support development
efforts as it would open the asterisk market to them. Anyone up for
creating a bounty-page for this?
I would be more than interested!
Anyone else? What would be the steps?
Aldo
[EMAIL PROTECTED] is believed to have said:
Hi,
which OSX softphone do you use that supports IAX2 protocol with Asterisk?
thanks
Mike
Hi Mike,
look for LoudHush on VersionTracker...
HTH
Aldo
___
--Bandwidth and Colocation provided by
[EMAIL PROTECTED] is believed to have said:
Can you explain this?
What country?
In this case it's not asterisk but the telco that has to do the Answer.
To every mobile? or just that provider?
Well,
it's funny because here, now (Italy; Telecom Italia PSTN calling Wind
mobile), I do get the
[EMAIL PROTECTED] is believed to have said:
I was not sure of the quality of both these hence i wanted some real
user experiences.
Thanks anyways...
Dan
Dan,
you're welcome...
While I can't speak for the quality of the Dock'n'Talk, as I did never
see one, the GSM gateway's audio quality is
[EMAIL PROTECTED] is believed to have said:
Hi,
Sorry for being very late on this thread but i am trying to make a
decision on which one to go for. Options are
1. Dock n Talk offered by Voxilla (USD139)
2. GSM Gateway by CyberTelecom (GBP60)
I'm having a TDM400P with 1 FXO FXS.
I'm
Benchev is believed to have said:
Thanks Aldo,
No I do not have a manual and I don't believe such a thing
exist. Actually, that GSM gateway is a Dock-N-Talk kind of thing
with the exception that the handset is imbedded, so pretty much
no need of a manual.
Is your Grandstream a HT-488? If so you
Benchev is believed to have said:
Hi,
Do you have any success receiving the caller id with your TDM400 FXO?
I have the same problem when I connect the GSM gateway to a SPA3000 FXO line
and thought this a Sipura's problem. On a phone connected to the GSM gateway
I can see the callerid, but not
Hi List
Is someone out there using one or more GSMgateway(s) from CyberTelecom ?
Me and some friends are interested in buying some of them, but before
we would like to ask, how the experiences are others have made.
e.g.
How easy to setup ?
How reliable ?
How's the voice quality ?
etc.
Any
Hi list,
any success trying to let internal calls ring differently than external
calls on a Grandstream BT102?
My settings, phoneside:
Default Ring Tone:system ring tone
x custom ring tone 1, used if incoming caller ID is *
custom ring tone 2, used if incoming caller ID is #
[EMAIL PROTECTED] is believed to have said:
Is it me, is it the evil fate, is it the BT102?
I even updated the firmware to Software Version:
Program-- 1.0.6.7Bootloader-- 1.0.1.0HTML-- 1.0.0.49VOC-- 1.0.1.0
TIA,
Aldo
The attempt to obtain custom ringtones I described in my
[EMAIL PROTECTED] is believed to have said:
Do you have the following set in your zapata.conf?
callerid=asreceived
Dear all,
I add my half cent on the subject.
I do have the following zapata.conf:
*
[channels]
usecallerid = yes
signalling = fxo_ks
callerid = A 2302
context =
Hello list.
I have a question about how to read the incoming calls' callerid on an
FXO interface of a TDM 400 analog card; (it's one of those RED modules).
Now -may this is the complexity adding step..- I have a GSM gateway
attached to this FXO thing; incoming calls are processed as they should.
[EMAIL PROTECTED] is believed to have said:
Hi,
Is there any syntax we can apply in the extensions to use
the anti-ex-girl(boy)friend technique to multiple callers without having
to replicate the lines?
I mean, can I write the following two lines in only one line?
exten=
[EMAIL PROTECTED] is believed to have said:
Bluetooth enabled phones to talk to Bluetooth headsets; I guess there's a
protocol for the phone to talk to any Bluetooth headset, no matter who made
it. This protocol would have to include something to allow voice to pass
from the phone to the headset
[EMAIL PROTECTED] is believed to have said:
[setup tool]
Sorry, I cannot answer that one. I don't know enough about these cards and
their drivers.
Armin,
thanks alot. One has to do some research and experimentation on his own
every now and then; and see if there is anything interesting that
[EMAIL PROTECTED] is believed to have said:
With BRIstuff you get to use ztcfg, etc.
Cannot say anything about mISDN, CAPI...
Francesco,
thank you; this is important to know
Aldo
___
--Bandwidth and Colocation provided by Easynews.com --
[EMAIL PROTECTED] is believed to have said:
While we are at the subject another couple of simple related question.
Are HFC-S cards active? I got one for a very low price, so that I
imagine it will be NOT the case...
No, these cards are passive. The protocol handling must be done by
the
[EMAIL PROTECTED] is believed to have said:
chan_capi does not set the NT-mode. Your cards driver need to do that.
E.g. for Eicon DIVA Server cards, you just set the '-x' option with divactrl
or set NT-mode in the config wizard.
chan_capi does not (need) to know anything about what protocol the
[EMAIL PROTECTED] is believed to have said:
The active cards do the ISDN protocol stuff on board, so the host CPU/driver
does not need to do that - better performance, less interrupts.
The AVM cards do not have such DSPs on board, so no echo-cancel.
But the Eicon DIVA Server cards do. They do
[EMAIL PROTECTED] is believed to have said:
We are considering it yes, but i don't know how hard or easy it would be.
I guess we will first try to make the other versions like we want them
to be and then start looking at other os'es.
Zoa
I have no clue, at all, too
So your plan makes
[EMAIL PROTECTED] is believed to have said:
Hey ho,
A few days ago we released the linux version of the phone, today we are
very happy to have the mac version ready for a little field test.
Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php
At the same time, we also
[EMAIL PROTECTED] is believed to have said:
Jason,
I'm sure these questions have been answered at some point, but I'm too new
to this stuff to know the right words to plug into the search function to
find what I need.
well, yes of course.
I have never touched Asterisk before, but have
[EMAIL PROTECTED] is believed to have said:
--
Message: 9
Date: Thu, 4 Aug 2005 07:43:51 +0200 (CEST)
From: Bastian Scholz [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] chan_capi upgrade
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Dear list,
today I installed a new asterisk machine, bound to replace my current pbx.
I am using a Fritz ISDN card; on the old machine with the drivers coming
together with the super-old rpm asterisk installation of SUSE 9.2.
The new machine is finally on asterisk 1.0.9, with chan_capi 0.5.4;
[EMAIL PROTECTED] is believed to have said:
The error is the same, afaik.
What I can't understand is why the make is entering in the directory '/
usr/src/linux-2.6.11.4-21.7-obj/i386/default'; I am by far not expert,
but I would expect it to go fiddle with a '586' directory.
Just a
[EMAIL PROTECTED] is believed to have said:
and watch linus himself rant about how this is incorrect to do (yet all
the distros do it) :P
Well, this is reassuring for a newbie like me.
Even the pros (as anybody building a distro ought to be, and most of the
times, really is) can do obvious
I know that this subject has been treated in the past!
As a matter of fact reading some old messages about compiling zaptel I
made a couple of tests after the first compiling failure to understand
why I can't compile on a specific machine, but I do not know how to
handle the results.
The machine
[EMAIL PROTECTED] is believed to have said:
I don't know if it is a phone like issue or not, but try the SPA-3000 setup
at http://geekgazette.com.
-Kerry
Kerry,
thanks for the hint. A first try did not get better results, but I was
doing it very quickly..
Aldo
[EMAIL PROTECTED] is believed to have said:
In the advanced options there are a few options for hang-up detection
including tone detection, and silence detection. They also have parameters
to adjust timing and sensitivy. IIRC, they are not enabled by default.
Nathan,
thanks: this is
Dear list,
I am playing with Sipura 3000 since last week.
Through the wiki pages I could get running it reasonably well.
My setup is that of a Sipura, linked with a local analog cordless phone,
a local PSTN line and the setup to link to an asterisk server located at
a remote static ip address.
[EMAIL PROTECTED] is believed to have said:
http://www.petitiononline.com/mod_perl/signed.cgi?IAXPhone
Sorry for the 170 or so who have already signed. This list supposedly
has 10,000 or more subscribers. 170 isn't very impressive. Please
sign!
Just signed; more hardware side support to the
[EMAIL PROTECTED] is believed to have said:
I installed Asterisk in a default way. I ran over many manuals and
FAQ's on asterisk.org. However, I found that many exaples included in
them were equipment-dependent. I do not know how to configure my
Asterisk for my X-Lite.
Is anybody willing to
Dear all,
I have tried to run an asterisk instance together with XLite on a single
machine (a PowerBook).
The intent is to take advantage of IAX connections to easily cross NATs
while traveling.
While the IAX setup proved 'easy', just having to fiddle a little with
working configs at both sides,
[EMAIL PROTECTED] is believed to have said:
Dear all,
I have tried to run an asterisk instance together with XLite on a single
machine (a PowerBook).
The intent is to take advantage of IAX connections to easily cross NATs
while traveling.
While the IAX setup proved 'easy', just having to
[EMAIL PROTECTED] is believed to have said:
It dosn't run under the mono framework. There, now you have an answer :-)
Oh, well: sad enough...
Thanks for the answer.
Aldo
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
[EMAIL PROTECTED] is believed to have said:
Hi Kong,
IAX2 support has been added in release 0.65 of IPSwitchBoard, which is ready
for download at
http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA, Zap is next
(I just need to get a Zap card).
Thorben
Hi Thorben,
IPSwitchBoard
[EMAIL PROTECTED] is believed to have said:
I am not sure that it will run on Mono, for now I only support it on
Windows. (I will test it on Mono later).
One thing (or so) at a time is indeed a good attitude ;-) ...
As soon as you'll feel to be the time to see what happens under Mono
please
[EMAIL PROTECTED] is believed to have said:
Suse 9.2 uses udev. Look for README.udev in you zaptel source directory and
follow the instructions.
Regards,
Alex
Thanks Alex!
Aldo
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
[EMAIL PROTECTED] is believed to have said:
I have a fairly current CVS build of asterisk running on SuSE 9.2. You
need to get rid of the stuff that gets installed with the system and
then install the zaptel stuff. Works fine for me, but I do get warnings
about unsupported modules and
[EMAIL PROTECTED] is believed to have said:
I've posted this question twice without a single reply. Does that mean no
one knows the answer, or no one cares to answer?
I've been having an issue with an IAX2 trunk setup in Asterisk. Setup the
trunk fine and it registers and works fine.
Of course I am not a kernel expert, so .. please be patient.
I am investigating on my zaptel/zapata problem.
As the main error message asterisk quits on mentions '/dev/zap/channel':
No such file or directory I went peeking over there.
[Asterisk Verbose Error
Mar 13 20:43:35 WARNING[5779]:
Hi list,
I am still attempting to start an asterisk 1.0.6 fresh installation.
There are some problems with the zap channel:
== Parsing '/etc/asterisk/zapata.conf': Found
Mar 13 10:49:38 WARNING[5278]: chan_zap.c:763 zt_open: Unable to open '/
dev/zap/channel': No such file or directory
Mar
Dear list,
I am trying to learn how to use Zap-things in Asterisk.
While loading Asterisk verbosely I get this error:
[chan_zap.so]Warning, flexibel rate not heavily tested!
= (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Mar 12 17:19:01 WARNING[5563]:
Dear all,
I have tried to compile * 1.0.6 (downloaded from the digium site, in the
right sequence - zaptel, libpri, asterisk) on two different machines
running SUSE 9.2.
The problem comes during some preliminary checks:
checking for ar... /usr/bin/ar
checking for tgetent in -ltermcap... no
[EMAIL PROTECTED] is believed to have said:
you need to install ncurses-dev
Thanks!
Aldo
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
[EMAIL PROTECTED] is believed to have said:
It's telling you that you have no curses devel package installed.
B
Thanks!
Aldo
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
[EMAIL PROTECTED] is believed to have said:
I have Asterisk set up at 2 offices, connected via an IAX trunk. My delema
is one person is always moving between offices. I have the dial plan set up
to ring phones at both offices but his voicemail box is at office A. His
phone at office A has the
Hi all,
sorry bothering again.
I am still stuck in compiling asterisk. Learning (or trying to) from the
first problem and first hint, when I got this error:
gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o
config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o
OK: I rushed for help too soon!
The analogy with the earlier problem (ncurses installed but not ncurses-
devel) struck back; I was lacking something related to openssl...
So I tried with openssl-devel and everything worked fine.
Sorry bothering the list,
cheers
Aldo
[EMAIL PROTECTED] is believed to have said:
Over here in Fedora Core land -
libssl.a and libssl.so live in /usr/lib are are installed there from the
RPM openssl-devel. Not sure how that translates to SuSE.
Don
Luckily quite closely to Fedora Core land!
Thanks
Aldo
[EMAIL PROTECTED] is believed to have said:
Hi, all
Got it to work finally. Thanks to all.
Had to add
[general]
externip=xxx.xxx.xxx.xxx ;ip address of your nat firewall (public ip)
localnet=192.168.0.0/24; the local subnet where the asterisk box is
Actually, I had 'externip' before,
[EMAIL PROTECTED] is believed to have said:
I am using BT102s and some generic voip phone. On the BT102 the transfer
button will put the call on hold and give you a new line to call an
extention with, however nothing happens when I call an extention. On the
generic voip phone the transfer
[EMAIL PROTECTED] is believed to have said:
I am using slackware 10.1 (kernel 2.4.29) and I am getting the following
when I issue gcc -v
Dimitris,
while I never compiled chan_capi I thought you would need a 2.6 kernel to
use it.
HTH
Aldo
___
Hi all,
is there a way to sense the automated announce messages that are sent by
cell phone operators?
I would like to switch to my own voicemail system if I dial a coworker's
cell ph. number and I am connected to the provider voicemail announce (or
if the cellphone is unavailable without
[EMAIL PROTECTED] is believed to have said:
On Wed, Feb 16, 2005 at 08:58:41PM +0100, Robert Rozman wrote:
Hi,
I'm trying to get capiECT working. I'd like to transfer call to
another ISDN
router connected extension and free channel from router to Asterisk.
I get incoming call on CAPI and
[EMAIL PROTECTED] is believed to have said:
Hey Everyone,
I downloaded and installed the X-Lite softphone the other day (the lite
version) and cannot seem to get it to work well.
Don't get me wrong, it registers with my asterisk server and everything
seems to work well, except the call quality
[EMAIL PROTECTED] is believed to have said:
http://fm.grandstream.com/gs/
Diego Aguirre
FWD# 459696
Thanks!
Aldo
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
[EMAIL PROTECTED] is believed to have said:
Diego Aguirre wrote:
I'm using 1.0.5.18 with no problems.
1.0.5.18 has an issue when registering (boot) and re-registering (after
register expiration, 1 hour) that appears an 403 for a minute on the
display (during this time the phone refuse
[EMAIL PROTECTED] is believed to have said:
In order to get the message button to work - programme it with the
extension number for your voice-mail. On your BT-100's phone web page -
it looks something like..
Voice Mail UserID:[300] (User ID/extension for 3rd party voice
mail system)
[EMAIL PROTECTED] is believed to have said:
Budgetone and MWI
The message button can be programmed to dial an extension that checks
voicemail
exten = 160,1,Voicemailmain(${CALLERIDNUM})
Thanks, this is what I was thinking about. Still, how do you get the BT
to dial 160?
In my Asterisk
[EMAIL PROTECTED] is believed to have said:
Other than doing an IVR type arrangement or a phone number dedicated to
VM access is there a way to do this? On my old POTS line I used to be
able to call my line and simply punch * during unavailable message
playback to go to the equivalent of
Aldo Bergamini is believed to have said:
This is how it was before:
; EXT. 2XXX
; generic dialer
exten = _2XXX,1,Dial(SIP/${EXTEN},20)
exten = _2XXX,2,Voicemail(u${EXTEN})
exten = _2XXX,3,Hangup()
exten = _2XXX,102,Voicemail(b${EXTEN})
exten = _2XXX,103,Hangup()
And this is how I changed
[EMAIL PROTECTED] is believed to have said:
I don't mean to be rude to everyone who responded to this question, but
I think that everyone is answering the wrong question. The point is that
the message waiting indicator doesn't light up, at all, ever. All that
happens when messages are waiting
[EMAIL PROTECTED] is believed to have said:
Hahawell the MWI is the blinking blue LCD. The message button
is reserved for future use Hang in there. There will soon to be some
No the message button call the number you configure in the web
interface. Presumably voicemail, but could be
[EMAIL PROTECTED] is believed to have said:
Ronald, it's the context listed in voicemail.conf (I got caught on this
as well)
I really wish Asterisk was better documented; it's bullshit the way it
stands at the moment.
Cheers,
Dean
Dean,
so if I have two contexts defined in voicemail.conf,
Since a couple of days I using an Asterisk server. I noticed something
obvious to anybody dealing with telephony since any longer time than
myself; echo is nasty...
Is it correct to say that the difference between a conversation between
SIP phone, Asterisk, an ISDN BRI line and a GSM phone is
[EMAIL PROTECTED] is believed to have said:
Hey Dan!!
Give us a clue as to what hardware/setup network provider you have there,
and we might be able to help :)
Paul
Hello Paul, hello everybody!
I have, too, an inbound call problem. I am using an ISDN Fritz Card PCI
2.00, together with
[EMAIL PROTECTED] is believed to have said:
Change the order of the lines and reload (or better restart Asterisk):
[interfaces]
msn=0221591030
incomingmsn=221591030
softdtmf=1
callgroup=1
context=from-chan_capi
devices=2
controller=1
Cheers, Philipp
Philipp,
THANKS!
I got past this stage
Sorry for the post with the bad subject
- AAB
[EMAIL PROTECTED] is believed to have said:
Change the order of the lines and reload (or better restart Asterisk):
[interfaces]
msn=0221591030
incomingmsn=221591030
softdtmf=1
callgroup=1
context=from-chan_capi
devices=2
controller=1
Cheers,
Dear list,
I am starting to setup an asterisk pbx, using a Fritz ISDN card through
chan_capi (0.3.5). The underlying OS is SUSE 9.2; I installed asterisk
with the RPMs supplied on the DVD.
While I can dial out (I had successful outside calls), through the ISDN
card, so far I could not answer a
[EMAIL PROTECTED] is believed to have said:
some time ago, I asked the list of a good book for learning ISDN and
SS7. I don't need to know how to write a channel driver or something; I
just want to know more about the possibilities and what's really sent
back and forth. I was told the book
Bruno Hertz is believed to have said:
Hi Aldo
don't know about Suse, but I have a working setup with asterisk 1-0
stable, chan_capi 0.3.5 and fcpci-suse9.1-3.11-02 on Debian Sarge,
though not prepackaged but all self compiled.
Looking at your log messages, chan_capi obviously is installed, but
Dear list,
I tried to install the CAPI support RPM from the SUSE 9.2 disks. YaST
tells me it is happy with the process. The Asterisk release I am using is
the one that comes packaged in RPM format, also included in the distribution.
Still starting asterisk with the usual asterisk -vvvc I see
[EMAIL PROTECTED] is believed to have said:
Hello ,
I'll just started with asterisk and I would liket to to dial between your
two phones with to cisco ATA 186 , but I have a problem
The two cisco ATA and the server in the same networks and i have the ring in
the phone but i'am not able
[EMAIL PROTECTED] is believed to have said:
Telnet uses TCP, SIP listens on UDP, use netstat instead.
B
Bob,
thanks for the hint! I should have imagined that SIP could not use a tcp
protocol...
Aldo
___
Asterisk-Users mailing list
[EMAIL
[EMAIL PROTECTED] is believed to have said:
Hello, I am a newbie with asterisk; I¥ve searching the mailinglist,
www.voip-info.org, isdn4linux web... But I don¥t know which isdn card to
buy.
I want the * box to be able to send faxes, and obviusly to send and receive
calls.
1) What do you recomend
93 matches
Mail list logo
