Hi again,
Kind of sheepish about asking for help, as I have only spent a day
banging my head off this...
I got my new Welltech 3701a, 1FXS,1FXO gateway.
I flashed it with what is seemingly the appropriate firmware (SIP
V1.04). This seems to have gone ok, and it is now registering both
ports ok with asterisk. For 1 minute I thought I was home free and and
everything was just going to work like magic. Wrong.
Problem #1) With my previous FXO (HT-488) I was using the following in
my dialplan:
exten => _NXXXXX,1,Dial(SIP/@2003,60,D(w${EXTEN}))
The above doesn't work for the Wellgate. So I tried:
exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@2003
This doesn't work either.
I am hoping to dial 7 digit numbers directly through the FXO.
Problem #2) Something is amiss with the DTMF. I can call in to
asterisk from my IAX phones and use Comedian mail fine (tones sent out
of band). If I dial the extension of my FXO I hear dial tone, but the
DTMF tones aren't heard, so I can't dial... Very odd, this also worked
perfectly with the HT-488
The "documentation" from Wellgate is a complete joke, with many choice
nonsense sentences.
If anyone has experience with this device, or ones that sound similar,
I would love some help and or ideas...
Thanks much,
Marty
Hello, Martin! At 02:50 AM 02/26/2006, you wrote:>I got my new Welltech 3701a, 1FXS,1FXO gateway.If you do give up with it (isn't Engrish documentation fun?), you may wish to take a look at the Sipura SPA-3000. I have one but haven't put it to use yet. I've heard *many* good things about it, though! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada
Martin Joseph
2006-Feb-27 01:44 UTC
[Asterisk-Users] Newbie config help? Wellgate 3701a (answers)
Short version:
Flash device with latest SIP firmware (currently 1.04)
Set "Network" (I am using the LAN port only) and "SIP"
config as
expected.
Set "Line configuration" so that the FXO is "hotline" to the
asterisk
extension you want to ring with incoming PSTN calls (mine is set to
2020).
Set "System configuration" so that the "keypad type" is
inband (rfc2833
doesn't seem to work?).
Change the "Routing Table" so that the default for "IP" is
set to FXO
for destination.
Click "commit data" and then the commit button.
Click "reboot" and then the reboot button.
Asterisk looks like this:
;
; SIP entry for user Wellgate (FXO)
[2003]
type=friend
secret=hushhush
dtmfmode=inband
auth=md5
host=dynamic
nat=yes
reinvite=no
canreinvite=no
disallow=all
allow=ulaw
context=autocontext
callerid="Alton Qwest Line"<2065551212>
;
; SIP entry for user Wellgate (FXS)
[2005]
type=friend
secret=Shhhhh
auth=md5
host=dynamic
disallow=all
allow=ulaw
allow=g729
allow=alaw
allow=gsm
allow=ilbc
context=autocontext
callerid="Alton Estates"<2005>
And the dialplan bit:
; Dial any 7 digit numbers through that plain old telephone network
exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@2003)
exten => _NXXXXXX,2,Hangup
;
Still a few minor issue 1) with double ringback on IAXCOMM, and one
with the beginning of audio being snipped on the FXS connected phone?
Not too bad though for a newb with a couple of 1/4 days :~)
I think it fixes my echo issue also. I can hear a sort of crackle for
the first 3 seconds of the call and then it's all good.
Hi,
I have another model 3702a (2FXS & 2FXO) voice gateway. You can implement
one-stage dialing on this device.
1. using "sip show peers" to make sure two ports (1fxo/1fxs) was
registered
to asterisk.
2. login 3701 web and change the defualt routing. The factory default is IP
-> FXS port. Change to IP -> FXO port
That's all. Just try Dial(SIP/${EXTEN}@your_3701_ip_address)
Kevin
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Martin Joseph
Sent: Sunday, February 26, 2006 3:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Newbie config help? Wellgate 3701a
Hi again,
Kind of sheepish about asking for help, as I have only spent a day banging
my head off this...
I got my new Welltech 3701a, 1FXS,1FXO gateway.
I flashed it with what is seemingly the appropriate firmware (SIP V1.04).
This seems to have gone ok, and it is now registering both ports ok with
asterisk. For 1 minute I thought I was home free and and everything was
just going to work like magic. Wrong.
Problem #1) With my previous FXO (HT-488) I was using the following in my
dialplan:
exten => _NXXXXX,1,Dial(SIP/@2003,60,D(w${EXTEN}))
The above doesn't work for the Wellgate. So I tried:
exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@2003
This doesn't work either.
I am hoping to dial 7 digit numbers directly through the FXO.
Problem #2) Something is amiss with the DTMF. I can call in to asterisk
from my IAX phones and use Comedian mail fine (tones sent out of band). If
I dial the extension of my FXO I hear dial tone, but the DTMF tones aren't
heard, so I can't dial... Very odd, this also worked perfectly with the
HT-488
The "documentation" from Wellgate is a complete joke, with many choice
nonsense sentences.
If anyone has experience with this device, or ones that sound similar, I
would love some help and or ideas...
Thanks much,
Marty
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