Displaying 20 results from an estimated 49 matches for "2fxs".
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2fxo
2009 Sep 08
1
Should digium build a 2FXO / 2FXS 4-port daughter board?
Please chime in if you've ever wished for digium to make a 4-port daughter
board with a combination of 2FXO AND 2FXS ports on the same card.
When using the 800 series cards, one must either choose 4-port permutations
of FXS/FXO, OR one must give up 2 valuable ports.
In other words, when you add ONE 100-series daughter board, you give up TWO
of your physical ports.
Is there a technical reason for the lack of s...
2014 Mar 20
0
[OT] Upgrading firmware AudioCodes MP-114 2FXS-2FXO from version 6.2 to version 6.6
Hello:
Is that possible?
Do I have to upgrade first to firmware version 6.4?
Does firmware version 6.4 exist?
Thanks in advance.
Regards.
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2007 Nov 27
3
Sip to ATA?
...net (currently satellite).
But, I cannot get much out of them, regarding how the phone service works. All I can get is I plug my existing phones and answering machines into the back of the "cable modem" and am good to go.
I am hesitant to believe that I can simply plug my TDM400P (2fxo/2fxs) into these (ATA ?) jacks and call it good.
Any insight? Am I better off ignoring their phone offering and setting myself up with an IAX or SIP provider? (and surplus-ing the card). I would end up needing more than their single line offering with a second line at $30/month (USD). Seems that mi...
2009 Nov 06
2
Question about callerid?
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with pretty good results.
Recently I added a callerid service to my landline (qwest).
I am using the audiocodes MP114 2fxo/2fxs gateway, which is an
outstanding piece of hardware once it's configured (lol).
Anyhow, I can see that the gateway is passing caller id info to
asterisk because the console will display something like:
[Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite:
Failed to authe...
2005 Aug 16
2
5 way calling?
I'm running RedHat 9 with a TDM400 (2FXO, 2FXS).
Before I implemented Asterisk, some users were using Bell services to
set-up 5 way calling: The user would set up a three way call on one
line, switch to the second line, set up another 3 way call and then
link the two lines together with the Flash key, thus establishing a 5
way call (the use...
2007 Sep 29
3
FAX detection not working
I am having a problem detecting incoming FAX. TMD22p (tdm400p 2 fxo, 2fxs)
As I understand it, I must have faxdetect = incoming to enable detection of the fax tone.
Then, I must have a [fax] context to pickup the line and send it to whatever extension the FAX device is on.
In my case, I ask it to answer immediately and do a distinctive ring (r3) to alert that is its a...
2010 Sep 07
2
5-7 second connection delay in outgoing FXO calls
I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys
SPA942 SIP phone and outgoing SIP and IAX routes.
When I dial local PSTN numbers from the SPA942 using the FXO channels I
observe a 5-7 second delay between when the PSTN number answers the call
and when Asterisk connects the call at my end. There's enough delay
time that I he...
2004 Feb 03
1
Mediatrix 1102 Auth
Hi all.
I'm evaluating a mediatrix 2fxs 1102.
seems great (it has also supervised transfer, that's
very needed in office environments and works well).
the only I thing I cannot make work is the auth
to my asterisk server.
If I don't set a password into the mediatrix and
*, I can call out, but still the registration goes wrong.
us...
2004 Dec 01
1
Micronet problem
Hello,
I connected a Micronet SP5014 2FXS + 2FXO gateway to the asterisk, the
problem is i can make call but can't receive calls. If i make a "sip
show peers" it shows the micronet is not connected to the asterisk.
Does anybody knows how to configure the micronet and asterisk to solve
this problem ?
Thank you
2005 Feb 09
1
TDM400P FXO lines problem
...days ago
we updated them to CVS version and today the problem reappeared. Average
time between failures seems to be around 20-30 days.
At first we thought it might be caused becouse we had 5 TDM400P cards in one
server. But the problems was also spotted in machines that had only one
TDM400P (2FXO/2FXS) card.
-- Mike
2005 Jun 07
1
error message: INIT: Id "s0" respawning too fast: disable for 5 minutes
I have set up asterisk@home <mailto:asterisk@home> with Digium TDM400P
2FXO/2FXS.
I am unable to seize my trunks from either soft or analog phones.
Inbound calls result in answer/disconnection.
I see the following error code on my asterisk server
INIT: Id "s0" respawning too fast: disable for 5 minutes
Does anyone have any suggestions for me?
I?d really appreciate...
2006 Jan 14
1
I need feed back on how an Aastra VentureIP 4FXO
...sterisk.
I'm thinking I'd need 2 to support 6-8 lines - Or suggest some other
equipment that will provide up to 8 fxo ports and connect to asterisk.
for future projects I'd also like something with 2 fxo ports and 4 - 5
fxs ports - I suppose a digium card would do fine for 2 fxo and 2fxs and
I could do a sipura 2002 for 2 more.
2006 Dec 11
1
Power requirements on the TDM-400 card
I have a TDM-400 from digium with 2FXO+2FXS ports. Any idea on how much power
will this drain from the 12 and 5 V connector when all ports are in use?
--
Gustavo Felisberto
(HumpBack)
Web: http://dev.gentoo.org/~humpback
Blog: http://blog.felisberto.net/
------------
It's most certainly GNU/Linux, not Linux. Read more at
http://www.gnu...
2011 Jan 14
1
5-7 second delay in connecting outgoing FXO calls
I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card,
SPA942 SIP phone and outgoing SIP and IAX routes.
When I dial local PSTN numbers from the SPA942 using the FXO channels I
observe a 5-7 second delay between when the PSTN number answers the call
and when Asterisk connects the call at my end. There's enough delay
time that I hear an add...
2008 Apr 02
1
FXS, Power and Sangoma
Hi
I've a Sangoma A200D with 2FXO and 2FXS. When using it with only
the FXO module, it's all good. But when I put in the FXS module and
connect the power, logs tells me not enough power.
> Mar 31 14:11:54 phone kernel: [ 4761.246931] wanpipe1: Module 1:
> Failed to powerup within 600 ms (8V : 72V)!
> Mar 31 14:11:54 ph...
2005 Mar 03
3
Problems dialing out - possible settings changes
...thing, but I could be searching for
the wrong info.
Second, I am new to asterisk and linux. My knowledge is Asterisk is
much better than linux, so excuse the ignorance in some areas.
Configuration:
Novell Linux Desktop
Asterisk 1.0.6
Intel CPU/MB
TDM12P digium card (I think that is it - 1FXO and 2FXS)
Problem:
My calls are infrequently successful. I have the dialplan to dial
_8NXXNXXXXXX and ignorepat => 8
When I dial 8206391xxxx, which is my cell... (and any of another 5
numbers I tried) About 1 out of 6 times it is dialed the connection is
made. The rest of the time I receive "the...
2012 Jun 18
1
TDM410 PTSN line setup with 1 analog phone
Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds
1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1
(not trying to use the gui, want to do everything by hand) with a TDM410 with
2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a
analog phone connected to port 3 (FXS). I compiled asterisk with asterisk
samples so I realize that may have messed me up.
This is all running on Ubuntu Server 12.04. I have been googling/researching
reading the book, etc. Every...
2006 Oct 30
0
Call from internal num. to VoIP gate
Greetings to All!
Help to solve a problem:
There is an asterisk and two VoIP a sluice (NSGate 800 2FXS 2FXO, NSGate
800 2FXS).
In sip.conf they are registered so:
[3301]
type=friend
host=172.222.135.11
username=3301
secret=0000
defaultip=172.222.135.11
dtmfmode=rfc2833
context=it
callerid="VoIPGate2Line1" <3301>
allow=g723.1
[3302]
type=friend
host=172.222.135.11
username=3302
sec...
2004 Dec 10
2
Asterisk 1.0.3 - Signaling on E100P.
Hello list ,
I?m putting to work a new asterisk box.
I?m running * 1.0.3 with (one )
Wildcard TDM ( 2FXS* 2FXO) and
(one) E-100P.
Both boards are working well. ztcfg
don?t show me any error.
zttool list both cards as "OK"
But when i run asterisk on verbose mode
i get those errors :
[chan_zap.so] => (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Regis...
2003 Mar 02
0
Malformed/Missing Contact field??
...means?
-- SIP/2113-945d answered SIP/111.222.96.14:5060
-- Attempting native bridge of SIP/111.222.96.14:5060 and SIP/2113-945d
-- Got SIP response 400 "Bad Request - 'Malformed/Missing Contact field'" back from 111.222.96.14
111.222.96.14 is a Cisco 1750 with a VIC-2FXS and a VIC-2FXO.
The call comes into the Cisco, gets sent to the Asterisk server,
the Asterisk server rings the FXS port on the Cisco. If I pick
up the ringing phone on the FXS port I get the above message.