Can anyone tell me if it is possible to invert the signaling bits on a T1
channel?
I need to emulate PLAR signaling in asterisk. EM seems to be an exact
match if reversed.
I need idle bits and seized
Thank you
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On Mon, Oct 01, 2012 at 06:44:22PM -0400, Pat Collins wrote:
Can anyone tell me if it is possible to invert the signaling bits on a
T1 channel?
I need to emulate PLAR signaling in asterisk. EM seems to be an
exact match if reversed.
I need idle bits and seized
Perhaps you could
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pat Collins
Sent: Tuesday, October 02, 2012 2:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DAHDI help please
Thank you for the reply!
So
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI help please
On Tue, Oct 02, 2012 at 11:22:31PM -0400, Pat Collins wrote:
Shaun,
To make more sense of the code, I changed
#define DAHDI_XBIT(3 2) to
#define DAHDI_XBIT(0)
Sadly
Try defaultuser=test instead of username=test
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Wolthuis
Sent: Thursday, October 04, 2012 11:01 PM
To: asterisk-users@lists.digium.com
Subject:
Check reinvite and NAT settings on the line as well as the SIP peers.
You can use a stun client from inside your network to see what’s going on with
the NAT
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Budinick
Sent:
)
-- Executing [444999@getpin:3] NoOp(SIP/testbridge2-0021, ) in
new stack //AND SO ON
Thank you!!
Pat Collins...
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} =
${CONF_PIN}]?getpin,good_exten,1)
Hope it helps,
Regards,
Bharat Lalcheta
On Thu, Apr 18, 2013 at 4:45 PM, Pat Collins drdialt...@optonline.net
wrote:
All,
Thank you in advance for any help.
I have a customer in need of a conferencing system
Thanks for the reply Doug!
How might I incorporate this into my dialplan?
Sorry, learning as I go
On a side note, any idea how to strip off the # at the end of a string?
${EXTEN:0:6} doesn't do the trick.
Is this even possible?
Pat...
From: asterisk-users-boun...@lists.digium.com
)
Hope it helps you out.
Regards,
Bharat Lalcheta
On Thu, Apr 18, 2013 at 5:36 PM, Pat Collins drdialt...@optonline.net
wrote:
Thank you Bharat.
Sadly, that made no difference.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
Hello group,
I put together a simple PHP based conferencing manager for Asterisk 11.3
I used ODBC MYSQL for conference IDs and PINs. All this is working as
desired but I would love to add an active conferences display to the front
end.
It seems to me that AMI is the way to go but I have no
Dialplan auto answer
; intercom
exten=_*,1,SIPAddHeader(Call-Info:
sip:xxx.xxx.xxx.xxx\;answer-after=0) ;xxx.xxx.xxx.xxx is the address of
your asterisk box
exten=_*,n,Dial(SIP/${EXTEN:1})
As long as your phones are compatible, this MIGHT work.
Worked for me. Sadly, I cannot
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, August 20, 2013 4:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cut off last character of EXTEN
Hello,
how can
2013, at 12:25, Pat Collins wrote:
Here ya go:
112233# use ${EXTEN:0:6})
123# use ${EXTEN:0:3})
123456789# use ${EXTEN:0:9})
I think 'variable length' implied 'unknown length'...
S
Yea, I realized that after I replied
Got ahead of myself again.
That was the only way I
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir
Mikhelson
Sent: Thursday, January 16, 2014 8:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 11
Hello all,
I've got some PHP code that opens an AMI socket and does a ConfBridgeList
for a specific bridge ().
This all works just fine but I need to filter the information displayed to
only CallerIDName so I can see a complete list of names of participants.
After days of googling and
Perhaps assigned as a test number somewhere along the line?
Are these ISDN, SIP, IAX calls?
There are MANY smart people on this list.
Maybe sharing the relevant configs and traces is a good place to start???
-Original Message-
From: asterisk-users-boun...@lists.digium.com
is sent from the CLI.
Is there any way to drop a channel when the peer using it disappears?
Googled every phrase I could think of. No luck.
Thank you!
Pat Collins
--
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On 04/11/14 15:11, Pat Collins wrote:
Hello group and thank you for the attention.
I'm using Asterisk 11.12 running on Ubuntu Server 12.04
We have an issue with channels remaining open after a SIP peer unregisters.
It seems that if the peer goes away before manually hanging up a call
Sounds like a job for TAPI.
Google TAPI for Asterisk or Asterisk TSP
I've been playing with SIPTAPI and it works pretty well. It's very simple to
install and set up.
Hope this Helps
PC...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Put qualify=yes in the peer definition in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Tuesday, December 30, 2014 1:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Have you looked into SLA? I have had good results with it. Will let asterisk
act like a key system.
Sent from Samsung tablet
Chris Gentle gent...@gmail.com wrote:
I need some advice on how to implement something in my dialplan.
Here's the scenario. A call comes in on my [incoming] context
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