This could be related to my not hearing voice through the phone over a
voice modem as well, which could answer both our problems if we have a
routine of things to check off.
thanks
On Sun, 2003-06-22 at 22:47, ayaz wrote:
> I am trying to test my dialogic card with asterisk , i have an E1 card
>
I am trying to test my dialogic card with asterisk
, i have an E1 card in asterisk and one dialogic(d300) in
another machine. Both are connected through a cross
cable.
Asterisk (digium E1) calls the
dialogic(d300) card to its answer demo. The answer demo shows that the call is
received b
Speaking of FWD, does anyone know about the status of their NAT server?
I get registered OK if I use their regular address, and the phones ring
and answer just like (I suspect) they ought to. But once the channels
are bridged, no audio passes, at least in my direction.
But if I use fwdnat.pulv
Someone said something about FWD registrations requiring the extended
syntax shortly, like this:
register => [EMAIL PROTECTED]:[EMAIL PROTECTED]/12345
At the moment, using the extended syntax doesn't work - I get
authorization errors. So stick with the old syntax:
register => 12345:[EMAIL PRO
On Mon, Jun 23, 2003 at 04:07:32PM +1200, Aaron Martin wrote:
> The hardware we are planning to use is:
>
> Micronet SP5050 FXO Gateway
> http://www.micronet.com.tw/Products/VoIP/SP5050.asp
>
> Micronet SP5100 IP Phone
> http://www.micronet.com.tw/Products/VoIP/SP5100.asp
>
I can't help you
The hardware we are planning to use
is:
Micronet SP5050 FXO Gateway
http://www.micronet.com.tw/Products/VoIP/SP5050.asp
Micronet SP5100 IP Phone
http://www.micronet.com.tw/Products/VoIP/SP5100.asp
We are hoping to use this hardware along with
AsteriskPBX to replace our aging PBX system.
Asterisk is becoming familiar to me, like getting a few friends after
moving to a different town, but I have had problems with modems from the
getgo on this.
Originally I wanted to do a voicemail system, but now I'm reading more
and more that excites me to use this setup and or add to it.
I sta
Hi all,
When I try to use SJPhone as a software SIP phone with Asterisk I have a
couple of problems.
When the phone register, I get the following line in the Asterisk console:
-- Registered SIP '' at 192.168.33.2 port 5060 expires 200
-- Got SIP response 481 "Subscription does not exist"
Hi,
I have such a problem too with a X100P and a phone connected to the line.
This is what I have seen in the Asterisk console.
-- Starting simple switch on 'Zap/1-1'
-- Executing Wait("Zap/1-1", "5") in new stack
-- Executing Dial("Zap/1-1", "SIP/home&SIP/dan|15|Ttm") in new stack
On Sunday 22 June 2003 02:18 pm, Tilghman Lesher wrote:
> On Sunday 22 June 2003 11:27, BK [address only for mailing lists] wrote:
> > Hi
> >
> > I followed the instructions on the Asterisk website for download and
> > building Asterisk. I checked out a fresh copy from the CVS tree as
> > described
On Sat, 21 Jun 2003, Ask Bjørn Hansen wrote:
>
> On Saturday, May 17, 2003, at 13:59 America/Los_Angeles, Brad Bergman
> wrote:
>
> > I'm having trouble with incoming calls that start to ring again a few
> > seconds after the local user hangs up. I get "Starting simple switch"
> > on
> > the F
Hi Chad,
if you are in the US you will need an ISN card that speaks NI-1.
The only CAPI cards that work with NI-1 are the Eicon Diva Server cards.
They are expensive active cards but the onboard DSP does echo
cancelation and some people already sucessfully use them in the US.
regards
kapejod
--
On Sunday 22 June 2003 11:27, BK [address only for mailing lists] wrote:
> Hi
>
> I followed the instructions on the Asterisk website for download and
> building Asterisk. I checked out a fresh copy from the CVS tree as
> described and that went smooth, but when I try to build as described,
> I get
asterisk -vvvcn | tee /tmp/log
CLI> sip debug
CLI> stop now
or
script
asterisk -vvvcn
CLI> sip debug
CLI> stop now
shell> exit
Martin
On Sun, 22 Jun 2003, destan wrote:
> Hi everybody,
> I want to read to debug messages and try to interpret them but they happen
> too fast, how can I log these g
Uriel Carrasquilla wrote:
They are being distributed by a couple of folks, one being ovislink. I
will
get you some numbers for contact on monday.
I just ordered my BT-100s with Ovislink, so I have got the details at
hand ...
Tel: +1 (626) 854-1805
Fax: +1 (626) 854-0835
they do have a websit
I visited the site but could not find prices or "buy" option.
I did come across an adapter for a analog phone. would it work the same way
as the SIP phone?
do you know the price?
does it have to be in the same LAN where * is located or can it access *
over the WAN with its own dynamic IP address?
Hi,
There is not a real FXO port. It just pass the FXS port to it when a PSTN
call is received and answered.
To make a little comparision is like a reverse voice modem (which only
aparently has one FXO port - the line and one FXS port - the phone and in
reality there is only a FXO port).
It will
Hi
I followed the instructions on the Asterisk website for download and
building Asterisk. I checked out a fresh copy from the CVS tree as
described and that went smooth, but when I try to build as described, I
get a truckload of errors and I have absolutely no clue what this all
means.
Can a
Hi All
Does someone tried Actiontec PCI Internet Phone Wizard or integrate to Asterisk
http://www.actiontec.com/products/voip/ipw_pci/ipw_overview_pci.html
http://www.actiontec.com/products/voip/ipw_usb/ipw_overview_usb.html
this stuff is like Quick cards BUT MUCH better it has FXO port and FXS
Just found it!
The problem was in gethostbyname ... simply
my hostname was not defined into /etc/hosts , so
sip was disabled on startup if bindaddr was = 0.0.0.0 .
Adding the hostname to /etc/hosts solved the issue.
Thanks a lot...
Matteo.
Il dom, 2003-06-22 alle 17:20, David P. Boswell ha scrit
On Sunday 22 June 2003 01:00 am, Chad Sawyer wrote:
> www.grandstream.com
>
> They are being distributed by a couple of folks, one being ovislink. I
> will get you some numbers for contact on monday.
Plz post them here...
Thx!
--
Steve
__
This sig is pendin
Hi all,
The lates version of Asterisk, using
busydetect=yes
busycount=5
in zapata.conf and [nl] as country works like a charm for my PSTN
connection.
I am located in Romania. It seems that the number of real heard busy tones
is busycount value -1 .
Have tried with busycount=5 and I hear 4 tones,
I have many dual homed asterisk boxes that work fine.
Kernel IP routing table
Destination Gateway Genmask Flags MSS Window irtt Iface
172.18.192.00.0.0.0 255.255.240.0 U40 0 0 eth0
10.10.0.0 0.0.0.0 255.255.0.0 U40 0
--On Sunday, June 22, 2003 14:38:20 +0200 Hervé Thibaud
<[EMAIL PROTECTED]> wrote:
i have an error when i start asterisk in :
chan_modem.so (Generic Voice Modem Driver)
-- Parsing "/etc/asterisk/modem.conf': Found
-- Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulates Modem
Driver)
Please
is there somebody who can help me with getting ADSI phones in Europe
I' am a little bit desperated. I need such a phone to play with * and adsi
features.
But i don't find a vendor who produce or a distributor who distribute such
phones in Europe.
I have found this link in
Le dim 22/06/2003 à 15:02, Andy Powell a écrit :
> >Andy, your update is
> >http://www.automated.it/guidetoasterisk.htm isn't it ?
>
> yes, same place, just added some extra notes in there (they should be obvious)
Yes my asterisk is on the internet gateway with sorewall (firewall) on
it and my st
>Andy, your update is
>http://www.automated.it/guidetoasterisk.htm isn't it ?
yes, same place, just added some extra notes in there (they should be obvious)
HTH
Andy
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Le dim 22/06/2003 à 12:18, Dan a écrit :
> exten => _8X,1,SetCallerID(${FWDUSERID})
> exten => _8X,2,SetCIDName(${FWDUSERNAME})
> exten => _8X,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> exten => _8X,4,Playback(invalid)
> exten => _8X,5,Hangup
It is better now, i try to call an ot
Hi list,
I am ready to implement VoIP for 2 of our offices. i do like to setup
asterisk box in each office (behind PBX) and use our ADSL links to provide
the tunnel. This will provide cheap telephone calls between offices.
I also like to setting PBX, so it can let incoming call-thourgh to asterisk
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Hi.
I've an asterisk box with 2 nics, one with a local
ip 192.168.1.xxx and the other with a public ip.
The default route is via the public ip route.
I need to use sip on both ends, since I have phones
on the lan and on the internet...
but.. if I set bindaddr=0.0.0.0 in sip.conf,
I had that error
>- Original Message -
>From: "Hervé Thibaud" <[EMAIL PROTECTED]>
>To: "asterisk-users" <[EMAIL PROTECTED]>
>Sent: Sunday, June 22, 2003 8:13 AM
>Subject: [Asterisk-Users] asteisk, sip & NAT
>hi
>My stations are behinds a firewall, the system is windows 2000 & 98, i
>use sjphone
>aterisk i
Hi everybody,
I want to read to debug messages and try to interpret them but they happen
too fast, how can I log these guys to a file, or is there a file like this
already?
I checked the /var/log/asterisk but there isn't much interesting there yet?
How can i turn on logging for SIP,IAX and other
Le dim 22/06/2003 à 10:16, Dan a écrit :
> Hi,
>
> Have you opened the port 5060 on your firewall? Then you need to open ports
> used for RTP, in order to have audio too.
yes i have opened 5060 & 5082 and it's OK if i call directly without
asterisk
> What do you exactly want to do?
> To call a F
Hi,
Have you opened the port 5060 on your firewall? Then you need to open ports
used for RTP, in order to have audio too.
What do you exactly want to do? To call a FWD user when you are connected to
your Asterisk box? To be called by an FWD user?
BR,
Dan
- Original Message -
From: "Her
hi
My stations are behinds a firewall, the system is windows 2000 & 98, i
use sjphone
asterisk is on the internet gateway where is the firewall Shorewall the
system is linux debian (sid) kernel 2.4.20
j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy)
to write my sip.conf but
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