Displaying 20 results from an estimated 20000 matches similar to: "Welltech FXS SIP registering with Asterisk"
2004 Jul 30
1
VoIP gateway (2 FXO, 2 FXS)
Does anyone know a good (and stable) voip gateway product with 4 ports
(2 fxo and 2 fxs), with the following requirements:
* being able to connect analog phones to the FXS ports, and communicate
over SIP with an REGISTRAR/PROXY server (SER in our case).
* being able to connect the FXO port to local office PSTN network, and
dial to that office pstn number and getting an internal dialtone, or
2005 Feb 24
2
Asterisk and Welltech USB SIP phone K1000A
Hi all
I'm fairly new to Asterisk, so be nice :-)
I was wondering if anyone has been able to get the Welltech K1000A USB
phone working on Linux. I see audio and HID drivers loaded when it is
plugged in to my Fedora Core 1 laptop, but that's about all that
happens. I've searched all the usual places (FAQ, Google, etc) but not
found anything helpful.
Asterisk is working fine with
2004 Jul 01
1
Help with Welltech 2FXO gateway, GS BT100 and Asterisk
Hi All,
I'm trying to configure 2 GS BT100 connected to asterisk and Welltech 2
ports FXO gateway. I configure WellTech 2ports FXO and GS BT100, both GS
BT100 can call each other without any problem but when I tried to call a
local extensions connected to my Welltech FXO gateway, I couldn't hear any
voice on both ends.
I would like to ask if anyone has ever encountered this kind of
2004 Oct 06
2
Working Wellgate *SIP* 38xx/35xx hardware anyone?
I'm loosing hair at cosmic speed now for the past 10 days.
Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to have
very buggy firmware possibly due to hastely done porting from H.323
firmware.
Is there anyone on this mailing list who was able to:
1. setup a 35xxA FXS with all ports authenticating properly with *?
or
2. setup a 38xx FXO to work as dial-in from pstn to
2007 Jan 26
1
WellTech 380x Gateway
Ok this is a simple question...
What has been your experience with the WellTech 38xx series (I'm looking
specifically at the 3802)
VoIP gateway? I'm looking for a good (and hopefully not too expensive)
VoIP/T.38 gateway for my office.
Asterisk intergration is not a major factor at this time but may be
later on. How well does it work? Is Echo
a problem? Do the T.38 capablities
2005 Mar 26
1
about sip and registering
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi folks,
I don't really understand the extension function on:
register => user:secret:authuser@host:port/extension
First question
- --------------
well, it's only local, or is important for authentication on external
sip server?
Example: I've one external sip account, the number is the URI also
(111), pass 'xxx'
I'll
2004 Aug 21
0
welltech fxo and *
hello everybody ,
i got a test setup with
lanphone 101 connected to *
and a welltech fxo 3802 also to *
the extensions are configure so that i can dial from the lanphone to
the fxo.
although once on the FXO and having the dialtone, no of my dtmf
dialing is being processed on the FXO. It keeps giving me the
dialtone constatly.
Everything is configured as outband DTMF (we
2008 Jan 14
0
Help needed for Fax2Email with Welltech FXO 3804
I have this in my extension.conf:
[incoming_28345474]
; 8862100 is the hotline number of the Welltech 3804
;
exten => 8862100,1,NoOp(${CALLERID(num)})
exten => 8862100,2,Wait(1)
exten => 8862100,3,Set(CALLERID(num)=${CALLERID(num)})
include => fax2emailstart
[fax2emailstart]
exten => 3000,1,SetVar(CALLEDFAX=${EXTEN}) ; me
exten => 3000,2,Answer
exten =>
2004 Apr 02
7
Welltech FXO: initial tests
Hi,
After a long way of problems (shipping, customs, etc) finally I got
Welltech working. Here below my comments.
- The documentation is poor and have errors
- The web configuration is not complete. However is useful for the basic
configuration parameters. The command line is necessary for modify all
parameters.
- The software upgrade is easy. Initially the gw came with H323, we
upgrade to
2006 Mar 07
1
PBX-VPN-SIP-Asterisk trouble
Hi all!
I have the following setup:
Phone lines -> traditional PBX -> Welltech 3802
-> VPN ->
Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone
There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2)
PBX extensions. Asterisk is a proxy here. Each device successfully
register itself. I tried the setup above with Linksys and Welltech
devices as well.
I setup
2015 Mar 15
0
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> Yes, I think the dial does get executed (sonny calling outbound
> 202-555-1212):
>
> core set verbose 3
> Console verbose was OFF and is now 3.
> -- Executing [912025551212 at from-internal:1]
> Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from
2004 Jun 01
1
Stuck SIP channels? -> SIP show channels
Hello all
I've discovered that SIP channels sometimes get stuck in *.
I've read some posts from Fri 29 Aug 2003 which mentions this issue, but
there doesn't seem to be any final answers
I don't know if this is related to the 0001604 bug?
Below is a list from one of the incidents:
I know the (d) means that it is scheduled for destruction but the 10.1.1.45
channel hasn't
2003 Nov 04
1
Flash hook -> SIP device
Hi there
I have a Welltech Wellgate SIP device and I want to be able to do a supervised transfer. I've read that in order to do that I have to use flash hook. The problem is just that I can't flash hook with this device.
I'm in contact with the developer of the SIP device but don't know what to tell him in order to get him to fix this.
What is happening when you flash hook, I
2015 Dec 02
4
Issues with Twilio number incoming call and context matching
Hello,
I am running Asterisk 13.6.0 in an AWS instance, and I set it up with
Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the
calls actually "reach" the PBX, but for some reason, they are not caught by
any of my extensions context.
Here's what I observe when I test this from a non-PBX connected E164 number
(a landline), say 555-666-1212. My Twilio number is
2006 Mar 23
1
wellgate 38XX FX & FXS voip gateways with outgoing call files
I am interested in the wellgate 38XX FXO and FXS gateways (and other
similiar units).
My question is can outgoing call files use these devices???
Can I fashion an outgoing call file with a channel like:
SIP/WellGate-1/5551212 (for the first port)
or SIP/WellGate-2/5551223 (for the second port)
TO these devices behave this way?
Of course incoming calls I dont see as a problem.
It's asking
2003 Jul 31
1
Zaptel cards, working FXS and SIP, no audio?
Greetings again all,
With the help of another list member I was able to get the my TDM400P
card working properly (and the PRI card loaded too for channels 1-24).
2010 Apr 26
0
DTMF from SIP phone to FXS/FXO
Hello,
I am having trouble passing DTMF digits from a Polycom 330 SIP phone to my FXS/FXO lines. I am running Asterisk 1.4.21.1
In sip.conf I configured dtmfmode=inband. RTP traffic (voice) goes perfectly from SIP to FXS, but in the SIP phone I only hear a continuous noise. However, when I press any digit in the pone (FXS) I hear the DTMF tone fine in the SIP phone (the noise goes away for as
2004 Oct 06
0
Anyone using the Micronet FXS/FXO devices w/SIP
I tried setting up a Micronet 4port 2 fxs 2fxo device using sip. I
found that I could get multiple fxs ports to register with *. I also
found that no matter what dialed on as an fxs user when dialing out it
would try to go out the fxo side of the micronet. I also had major
problems with these devices behind nat. I was having so much fun with
these little boxes that I put them back on the shelf
2008 May 22
0
SIP configuration issues
Apologies if this is a repeat: I trawled through the archives and couldn't
find a reasonable answer, so I'm asking here. I have an Asterisk install
connecting from behind a NAT device (DSL modem) to a SIP proxy (in my case,
Broadvoice). I have an sjphone softphone on a Windows PC also behind the NAT
device that connects to the Asterisk install, and using this setup I've been
pretty
2005 Aug 25
0
Internal FXS to SIP problem
I've just setup a new asterisk box (cvs HEAD) with a digium tdm411 and
a couple computers with eyebeam. I have one small. I cannot call the
eyebeam clients from the phone connected the fxs port. I can call the
phone from the eyebeem clients. And, I get both the fxs phone and
eyebeam clients to ring when a call comes in through the fxo port.
I have been trying to get this straightened out