All,
I'm running CVS-HEAD as of 15thJune with an x100p and the x100p callerid
patch. I'm trying to use app_sms to recieve sms to my landline but get
the following response.
Any ideas.
-- Executing NoOp(Zap/1-1, Testing without Answer 08005875290) in
new stack
-- Executing
Tony,
I'm havin a similar issue i'm in the UK using x100p with the patch for
CID and get the following. Any ideas
Executing Goto(Zap/1-1, sms-in|s|1) in new stack
-- Goto (sms-in,s,1)
-- Executing SMS(Zap/1-1, default|a) in new stack
-- SMS TX 93 00 6D
-- SMS RX 93 00 6D
--
.
Jon
Tony Hoyle wrote:
Jon Creasey wrote:
Tony,
I'm havin a similar issue i'm in the UK using x100p with the patch
for CID and get the following. Any ideas
Did you change line as mentioned earlier? Without that patch incoming
SMS won't work at all.
You might need to use an even higher
All,
What i'm trying to do is setup a windows DUN connection via my asterisk box
and over PSTN or VOIP to my work. What I hoped i'd find was a vitual modem
driver for windows 2000 that wouldtalk over sip to my asterisk box and then
act like a normal modem so I can dial out from that to our RAS
Quoting Jon Creasey [EMAIL PROTECTED]:
All,
What i'm trying to do is setup a windows DUN connection via my asterisk
box
and over PSTN or VOIP to my work. What I hoped i'd find was a vitual
modem
driver for windows 2000 that wouldtalk over sip to my asterisk box and
then
act like a normal
shows that during the initial exchange (a
number of times), and not thereafter. That shouldn't be the case.
Regards,
Steve
Jon Creasey wrote:
All,
Using CVS-03/05/04-17:17:29 asterisk
Spandsp 0.0.1f
Getting the following when faxes come in on x100p
Receiver's minimum scan line time
All,
Using CVS-03/05/04-17:17:29 asterisk
Spandsp 0.0.1f
Getting the following when faxes come in on x100p
Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
R8x15.4lines/mm OK
Minimum scan line time for higher resolutions: T15.4 = T7.7
DIS: 80 00 ce f0 80 80 01
T4 timeout in
Would it be too much trouble to put a few details of the setup together
hardeware etc and reply back to the list so we can all get some ideas of how
it can work in this sort of environment.
Many Thanks
Jon
- Original Message -
From: Jared Smith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Using DIAX softphone which seems to be working OK can get to VM/echotest etc
in the demo context
Am trying to setup FWD but get the following problems
Can hear it ringing when dialing FWD no 612 for time. Connects but no sound
from remote end.
Does anyone have any suggestions.
Softphone on
Hi all,
New user to asterisk having just got it compiled and installed.
Running with no digium hardware (yet) and no soundcard in asterisk box.
Problem is using the sample configs with a sip phone added as follows
[2203]
type=friend
username=2203
secret=2203
host=dynamic
defaultip=192.168.0.2
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