Hi Dennis,
You need to add immediate=no before your channel assignment; asterisk
will then give you a dialtone when you pick up the handset. Also the
context uniware_sendfax must have the pattern match and associated zap
dial that you need.
HTH.
Cheers,
Paul
www.austechpartnerships.com
t)
Hi Dennis,
Yes, you've got it correct. Glad its working for you.
Cheers,
Paul
www.austechpartnerships.com
t) +61 (0)3 9221 0888
SIP) [EMAIL PROTECTED]
IAX) [EMAIL PROTECTED]
IAXtel) 1700-482-8273
ATP Centrex)
Dennis wrote:
Hi,
Thanks for that Paul, it has solved that problem at
Hi Dean,
We at ATP have a range of resellers/integrators on our system to provide
solutions around Australia. Get them to contact us, and we'll put them
in touch with the nearest integrator with the correct skillset.
Cheers,
Paul
www.austechpartnerships.com
t) +61 (0)3 9221 0888
SIP) [EMAIL
Hi Alex,
Have you checked that your jumper setting on the card has been shorted
for E1. Its open by default to T1 - which overrides your zaptel.conf
settings.
Cheers,
Paul
Alex Barnes wrote:
Dec 28 12:51:55 caudi_apx1 kernel: TE2XXP: Launching card: 0
Dec 28 12:51:55 caudi_apx1 kernel:
I am sorry, you lost me here? You mean set rxflash to the max and
flash to the min time? What times should I use?
Currently I have:
pulsedial=no
flash=100
rxflash=100
Hi Brian,
flash=80
rxflash=120
Cheers,
Paul
--
www.austechpartnerships.com
t) +61 (0)3 9221 0888
SIP) [EMAIL PROTECTED]
Brian May wrote:
Hello,
How do I get the recall button working on a phone attached to a TDM400
FXS port using Asterisk?
I did a web search, and found people with exactly the same problem,
but no solution.
I suspect the timing is set for American standards, is it possible to
get it to work
Hi Brian
Ideally I would like to continue using the pre-built binary for
Debian. If possible.
Unfortunately - the only way is to submit a patch
Also, I would assume that rxflash and/or flash in zapata.conf does
the same thing, but so far I haven't had any luck. As such, I am not
entirely
Christopher Lee wrote:
Out of interest is there any estimated date for the TDM400 FXO modules
receiving A-tick certification?
And has anyone compared the FXO modules with the X100P on Australian
exchanges/equipment? Do they perform any better than the X100?
Cheers,
Chris Lee
From: Erik Barker [EMAIL PROTECTED]
We are currently using Polycom IP600 VOIP phones for our office which
are capable of handling 2 calls per SIP registration. What we're finding
is when staff are on the phone, Asterisk will pass them a second call
which will show up on their display, and an
From: Owen Kelso [EMAIL PROTECTED]
Sent: Sunday, January 11, 2004 10:07 AM
Subject: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)
On the NAT'ed side I have the BudgetTone set up to use STUN and ports 5060
for SIP and 19000 for RTP. The firewall that performs NAT forwards ports
5060
- Original Message -
From: Martin [EMAIL PROTECTED]
mkdir -p /sbin
install -m 755 ztcfg /sbin
make: install: Command not found
make: *** [install] Error 127
[EMAIL PROTECTED] zaptel]#
Why won't zaptel make install ?
Martin
Martin,
Looks like somewhere along the line, you
- Original Message -
From: Michael [EMAIL PROTECTED]
Sent: Thursday, January 08, 2004 10:40 PM
if you are on a call on the Budgetone 101 and a 2nd call is received,
instead
of a call waiting beep being played, it rings on the handset speaker!
which
makes it almost impossible to
- Original Message -
From: John Coll [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 05, 2004 9:07 AM
Subject: [Asterisk-Users] Newbie - MWI
Sorry for the partial post a moment ago
With help I got two phones communicating - PCMA/PCMU was the problem.
Next stpe is to
- Original Message -
From: Stephen J. Wilcox [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 22, 2003 12:54 AM
Subject: [Asterisk-Users] Callwaiting / limits?
Hi,
I'm using grandstream phones, when on a call and a second call comes in
the
call waiting indication is
- Original Message -
From: Michael T Farnworth [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 21, 2003 4:31 AM
Subject: Re: [Asterisk-Users] Best SIP PHones to buy ?
We have bought around 30 Grandstream phones, both BT101 and BT102. In
general the phone is
- Original Message -
From: Balaji NJL
To: [EMAIL PROTECTED]
Sent: Monday, December 15, 2003 8:47 AM
Subject: [Asterisk-Users] unable to configure my Grandstream phone
snip
Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2
WARNING[5126]: File chan_sip.c, Line 1954
- Original Message -
From: Paul Lambert [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: Paul Liew [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 4:16 AM
Subject: Re: [Asterisk-Users] incominglimit stuck in app_queue
I've seen this same thing. But it doesn't happen only for phones
- Original Message -
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 29, 2003 3:34 AM
Subject: Re: [Asterisk-Users] call waiting disable in sip
what would happend if all operators are busy? would app_queue exit?
would it schedule the call to
callwaiting=no is not supported by chan_sip. Call waiting
enabling/disabling is a function of SIP phones. Unfortunately, GS does not
support disabling call waiting as yet, so I've had to put in a patch to
overcome the problem. Look under
http://bugs.digium.com/bug_view_page.php?bug_id=408. You
Hi Mick,
It's going to be hard for anybody here on the list to help you, unless you
are more specific, ie, what you did exactly to get a crash, and console
output (with verbose set) debugs, logs (under /var/log/asterisk) and some
configuration files. We'll be in a better position to help you then
Also, check my patch http://bugs.digium.com/bug_view_page.php?bug_id=408
which does fix incominglimit/outgoinglimit. Stops callwaiting on sip phones.
Hope that helps.
Paul
- Original Message -
From: David Gomillion [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 18,
Is it possible to incorporate iLBC codec, some hotels only allow 28.8
dial-up links, and then your product will be really useful on the
road
How _does_ * work on dialup? I have never tried. I know you have an
immediate 200-300ms lag but how is it otherwise?
I've used a GS
Sorry to do this to the list, but I have no choice .
Walker,
I've been trying to send you an email off-list for the last couple of weeks,
but one of my mail-hops is failing, do you have alternative address that I
can try ???
Paul
___
Asterisk-Users
Hi
?php
// From Kapjod's sample..
ob_implicit_flush(true);
set_time_limit(0);
$err = fopen(php://stderr,w);
$in = fopen(php://stdin,r);
$out = fopen(php://stdout,w);
//This works..
fputs($out, Verbose \Calling phone\n);
// This doesn't
fputs($out, exec(Dial(sip/2012)\n);
fclose($in);
- Original Message -
From: Billy Huddleston [EMAIL PROTECTED]
how could you do this with sip and VOIP?
From: Steven Critchfield [EMAIL PROTECTED]
You want to look into call groups and pickup groups. To pickup the call
you use *8#.
from
Hi Dan,
Nice goingsome testing feedback. Testing your new client with
voicemail - after entering password (4 digits), 1 for new messages, then no
further digits can be sent. So far everything else OK.
Regards,
Paul
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL
at 10:28:32AM +1100, Paul Liew wrote:
Hi all,
Having fixed my problems with the call waiting ringing on the GS phones,
I needed to extend that with a campon facility (available on some legacy
systems - sort of callwaiting without phone ringing). I've managed to
implement that by adding
I am thinking of coding a solution using variables, Cut, and ChanIsAvail.
here is what i'm thinking of doing
Create a variable that contains the string SIP/gs1SIP/gs2SIP/gs3 ...
etc
check each phone with ChanIsAvail, and use Cut to remove its
representation
in the string (if its not
Sean Rodger wrote:
Is there anyway to turn off the call waiting beep in the grandstream
and/or
cisco ata186?
I have a dial statement in my extensions.conf that rings 5 phones at the
same time by combining them with the in the dial statement.
i.e.) exten =
Michael,
A couple of things - having a quick look at the app_ChanIsAvail code - it
seems that it is designed for Zap devices, so using them on any SIP phones
would not provide the expected result. Secondly, which SIP phone are you
using, I can't put calls on hold and make further calls without
Hi all,
Having fixed my problems with the call waiting
ringing on the GS phones, I needed to extend that with a campon facility
(available on some legacy systems - sort of callwaiting without phone ringing).
I've managed to implement that by adding/modifying app_queue.c. Basically, when
Michael,
I've added a patch a week ago on to bugtracker to fix this - feel free to
try it and let me know
http://bugs.digium.com/bug_view_page.php?bug_id=408
Paul
- Original Message -
From: Michael Ulitskiy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 29, 2003
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 22, 2003 11:10 PM
Subject: Re: [Asterisk-Users] Call Waiting on SIP phones
Subject: Re: [Asterisk-Users] Call Waiting on SIP phones
Paul, I applied the patch successfully last
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 22, 2003 5:50 AM
Subject: Re: [Asterisk-Users] Call Waiting on SIP phones
Paul, I'm getting a patch error when I diff to the chan_sip.c that I just
got from CVS this morning. It
Hi All,
This is the first time I'm submitting a patch, and I hope it fixes more than
it breaks. I'm putting it here, since John Todd mentioned a while ago about
the heavy load Mark and crew have at Digium (doing such good work), so I
thought all of us could test this first, and if ok submit for
Sorry, to repost - but I left a "/*" comment - here
it is again
Paul
--- chan_sip.c.save
2003-10-20 21:51:52.0 +1000+++ chan_sip.c 2003-10-21
09:26:41.0 +1000@@ -959,7 +959,9
@@
return 0;
} switch(event)
{+
/* Incoming and outging affects the inUse counter
*/
case
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 21, 2003 10:29 AM
Subject: Re: [Asterisk-Users] Call Waiting on SIP phones
Paul -
A few questions and comments:
1) So, does this also make incominglimit and outgoinglimit work
If you had a look under the help as the prompt said and entered help
show - you would have found that it is show codecs
Paul
- Original Message -
From: Aaron Martin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 20, 2003 11:56 AM
Subject: Re: [Asterisk-Users] use of SIP
Dave,
After the system boots up, you check to see which modules are loaded by
doing a lsmod. zaptel and others that you need should be listed. If not
you can manually add modprobe zaptel and the other drivers into your
rc.local file.
Paul
Just before the Logon prompt appears on boot I see a
The number of digits that your telco sends to you is a configurable figure
(at least it is here in Aus). The example assumes that the telco is sending
you the last 4 digits.
Paul
Example: 456-7000 is your main number and you have 7001 to 7099 as DIDs:
exten = 7000,1,Goto(AutoAttendant|s|1)
You can also use the AGI interface function RECORD FILE and specify a max
record duration of 5s and silence detection of 1s. Time the duration of the
call to asterisk - if its longer than 1 second you know you've got voice. If
you need to check for voice over a longer period of time - repeat the
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