On Sun, 2010-01-17 at 23:28 -0800, Lee Howard wrote:
Kingsley Tart wrote:
Jan 14 12:44:49.39: [ 3403]: -- [9:AT+FRH=3\r]
Jan 14 12:44:56.39: [ 3403]: -- [0:]
Jan 14 12:44:56.39: [ 3403]: MODEM Empty line
Jan 14 12:44:56.39: [ 3403]: MODEM TIMEOUT: waiting for v.21 carrier
Jan 14
Kingsley Tart wrote:
Do you know what I should look at next, or how to get more diagnostics
somehow?
Record the fax using the record option in your iaxmodem config file.
The files will be put into the /tmp or /root/tmp folder. You can play
them back with Audacity.
This way, you'll be
On Mon, 2010-01-18 at 07:03 -0500, Doug Lytle wrote:
Kingsley Tart wrote:
Do you know what I should look at next, or how to get more diagnostics
somehow?
Record the fax using the record option in your iaxmodem config file.
The files will be put into the /tmp or /root/tmp folder.
Hi All,
Maybe someone knows this, im using dahdi in combination with a TDM400,
where 2 analog PSTN lines are connected.
The weird thing is tho that when someone calls the analog lines it goes
perfectly fine, the line comes in and all works ok.
Except:
Sometimes the callerid from the caller is not
ev...@disruptor.nl wrote:
Except:
Sometimes the callerid from the caller is not the complete number, but
Just a guess, try an Answer(1)
Doug
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Stupid that i didnt think of that :)
i'll try that:)
Regards,
Evert
ev...@disruptor.nl wrote:
Except:
Sometimes the callerid from the caller is not the complete number, but
Just a guess, try an Answer(1)
Doug
--
_
Thank you for advice.
Do you get the same results if you use:
iax2 test losspct x
Where x is the loss percent you'd like to test?
Yes, I did it.
On CLI show:
VvvvLvvvLLvv
vvLvvLvvv
Ok just try'ed the Answer(1) but nope, still its only a few numbers, still
sometimes it goes ok, and sometimes its just a few numbers.
Regards,
Evert
Stupid that i didnt think of that :)
i'll try that:)
Regards,
Evert
ev...@disruptor.nl wrote:
Except:
Sometimes the callerid from the
Try Answer(5). Don't know how long the Dutch system takes to connect a
call, but it should not take over 5 seconds. If that doesn't solve it,
timing is not your issue.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Ok try'd 5, but still its missing numbers sometimes.
Regards,
Evert
Try Answer(5). Don't know how long the Dutch system takes to connect a
call, but it should not take over 5 seconds. If that doesn't solve it,
timing is not your issue.
-Original Message-
From:
In theory, Answer(5) should be the same as this
- exten = s,1,Answer
- exten = s,Wait(5)
I'd try those two lines and
- exten = verbose(ID from telco ${CALLERID(num)})
This should tell you what is coming in from Telco. Just out of curiosity,
do you have a manual ID box/phone to verify the
You may have a gain issue. Since the Caller ID information on an
'analog' line is FSK it is sensitive to distortion. How are the quality
of your lines, do you have a hum or wicked echo? Run fxotune if you have
not done so already.
The Answer() that you added would apply on PRI circuits that send
Hey Danny,
Yes there is an analog phone parallel to the connection going into the
digium card.
And it shows the numbers correctly.
Try'd the lines you mentioned below also, with exactly the same result still.
Regards,
Evert
In theory, Answer(5) should be the same as this
- exten =
On Mon, 18 Jan 2010, ev...@disruptor.nl wrote:
Hi All,
Maybe someone knows this, im using dahdi in combination with a TDM400,
where 2 analog PSTN lines are connected.
The weird thing is tho that when someone calls the analog lines it goes
perfectly fine, the line comes in and all works ok.
I haven't try'd fxotune yet, going to lookup how i need to run that properly.
Thanks for the advice!
Regards,
Evert
You may have a gain issue. Since the Caller ID information on an
'analog' line is FSK it is sensitive to distortion. How are the quality
of your lines, do you have a hum or
Ok after the fxotune, it still does it.
i hear no weird hum or wicked echo's
Regards,
Evert
You may have a gain issue. Since the Caller ID information on an
'analog' line is FSK it is sensitive to distortion. How are the quality
of your lines, do you have a hum or wicked echo? Run fxotune
At 09:13 AM 1/18/2010, you wrote:
Maybe someone knows this, im using dahdi in combination with a TDM400,
where 2 analog PSTN lines are connected.
The weird thing is tho that when someone calls the analog lines it goes
perfectly fine, the line comes in and all works ok.
Except:
Sometimes the
Hey Gordon,
Those settings are set to =nl in my config for dahdi.
Im assuming that would be correct :)
Regards,
Evert
loadzone=uk
defaultzone=uk
Work out what's right for .nl and it'll be a good start.
Gordon
--
Ira wrote:
At 09:13 AM 1/18/2010, you wrote:
Add a WAIT(1) as the first line of the incoming context and see if
that helps.
Answer(1) is the same as WAIT(1)
Doug
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Hey Ira,
It seems after a several testing, that the wait(1) seems to solve the issue.
Only now weirdly enough the phone keeps ringing if the caller hangs up
before i picked up the phone (pstn call)
Regards,
Evert
Add a WAIT(1) as the first line of the incoming context and see if that
Hey Doug,
According to the commands its:
Answer(ms)
Wait(s)
So it seems not the same.
Regards,
Evert
Ira wrote:
At 09:13 AM 1/18/2010, you wrote:
Add a WAIT(1) as the first line of the incoming context and see if
that helps.
Answer(1) is the same as WAIT(1)
Doug
--
Not necessarily unless WAIT(1) does an Answer if not already done.
According to CLI Show application answer, answer will pick up the line on
the first instance, and just wait on subsequent ones. Wait only waits...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
ev...@disruptor.nl wrote:
Hey Doug,
According to the commands its:
Answer(ms)
Wait(s)
So it seems not the same.
I should have been more specific. They both will wait 1 second.
Doug
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On Mon, 18 Jan 2010, Doug Lytle wrote:
Ira wrote:
At 09:13 AM 1/18/2010, you wrote:
Add a WAIT(1) as the first line of the incoming context and see if
that helps.
Answer(1) is the same as WAIT(1)
Not in my book...
Answer will answer a call, then optionally wait.
Wait will simply wait
At 11:52 AM 1/18/2010, you wrote:
It seems after a several testing, that the wait(1) seems to solve the issue.
Only now weirdly enough the phone keeps ringing if the caller hangs up
before i picked up the phone (pstn call)
It's what I used years ago when I first installed Asterisk. If the
phone
At 11:54 AM 1/18/2010, you wrote:
According to the commands its:
Answer(ms)
Wait(s)
So it seems not the same.
I should have been more specific. They both will wait 1 second.
While I've never used answer, I assume it picks up the line which
might not be what is wanted, it's not in
Hi:
I Bought TDM2400P ,with 24 FXO ports , I installed the asterisk 1.4.28
and dahdi 2.2.0 and then i compiled them and configured all, i pluged into 8
pstn line into the Rj connector and then i got messages on asterisk console
that alarm cleared on channels 1-8 , at this step everything
On Mon, 18 Jan 2010, Gordon Henderson wrote:
Phone lines make out of liquorice or something ;-)
Been hitting the sake early today?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice:
On Mon, Jan 18, 2010 at 08:32:26PM +0100, ev...@disruptor.nl wrote:
Hey Gordon,
Those settings are set to =nl in my config for dahdi.
Im assuming that would be correct :)
Those are not related to caller ID detection. Most of it is done by
Asterisk, anyway.
--
Tzafrir Cohen
On Mon, Jan 18, 2010 at 08:13:28PM +0100, ev...@disruptor.nl wrote:
Hey Danny,
Yes there is an analog phone parallel to the connection going into the
digium card.
And it shows the numbers correctly.
Try'd the lines you mentioned below also, with exactly the same result still.
Best thing
On Mon, Jan 18, 2010 at 07:15:28PM +, Gordon Henderson wrote:
Dutch caller ID - it's weird. Seems to be transmitted in DTMF before the
line rings, so answering it isn't going to help at all, and by the time
you get the ring, it's too late anyway.
I can't find anything definitive (and
Hey Tzafrir,
In my first mail i already said that i receive the clid strings, only
sometimes numbers were scrambled aka missing numbers from the complete
number.
Regards,
Evert
Try'd the lines you mentioned below also, with exactly the same result
still.
Best thing to do to provide more
On Mon, 18 Jan 2010, Steve Edwards wrote:
On Mon, 18 Jan 2010, Gordon Henderson wrote:
Phone lines make out of liquorice or something ;-)
Been hitting the sake early today?
Well, it's 9:30 pm where I am, and being scottish, it might be something
else, I'd prefer, however that was intended
I have an Audiocodes MP-114, in sip.conf I have two entire for PSTN line:
[pstn-5665] ; incoming/outgoing calls on FXO port 5665
type=friend
secret=
insecure=invite
username=fax-5665
host=dynamic
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
What user are you running Asterisk as?
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Cheers,
Matt Riddell
Managing Director
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http://www.venturevoip.com/exchange.php (Full ITSP Solution)
http://www.venturevoip.com/st.php
hi.
What user are you running Asterisk as?
I tried 2 patarn.
First , I worked asterisk as 'asterisk', and tested.
But jitter and PLC didn't work correct.
So I thought it may be caused permission problem,
and made a new system working asterisk as 'root'.
Now I tested as root.
And same
hi,
i try to convert wav file to gsm format.use following commands;
sox net263-welcome.wav -r 8000 -g -c 1 net263-welcome.gsm resample -ql
the file is located in /var/lib/asterisk/sounds/net263
but cant' play.do you know what's wrong?
-- Executing Playback(SIP/1001-0091,
Hi Don and others.
Finally, we've set up our Asterisk with ISDN service. At the edge of our
network, we can see all three numbers we are interested in as follows.
D1 : L3 TX CREF=0004 IE[05]=CALLGNUM (NumType=National
NumPlan=ISDN/Telephony[E.164] PresentInd=Allowed ScrnInd=NetworkProvided
Hello,
I need to implement B2bua in Asterisk by using java fastagi could any one
give me any idea?
Thanks
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Ahmed Magdy Mahmoud
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