it into the guest account.
iv posted a bug with a bit more detail but it was closed as a
configuration issue (which i suppose it is...)
http://bugs.digium.com/bug_view_page.php?bug_id=0003621
Morgan Gilroy,
Telappliant Support
___
Asterisk-Users
To get around this i updated CVS HEAD and changed the sip entity
from
type=user to type=peer (yes peer!) (type=friend works too but im
making
a point) the client now must register to set his outbound caller*ID
Number.
Yes, that is normal. SIP has difficulty separating the remote
Yes, exactly (and there will be other settings as well, to identify
the
type of peer (network, trunk, endpoint) for other reasons).
cool, I really should read the lists more :)
That's coming too, but in a different way. Actually if your remote
peer
can send you Remote-Party-ID
help
I just want a list of commands, if this mail shows in the list,
sorry, my bad.
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even while im still 'in' the DIAL app and the call continues on just
fine.
Iv looked through all my scripts and cant see anything to cause this.
Im pretty new to asterisk so I don't know what to do now..
Morgan Gilroy
Support
i2 Networks Ltd
tel 0871 717 7540
fax 0871 717 7541
: Re: [Asterisk-Users] CDR shows billsec=12 for all bridged
calles.
On 2 Jul 2004 at 16:17, Morgan Gilroy wrote:
Can someone help me, im using latest CVS, asterisk and cdr_mysql,
when I make a bridge call (using .call files in outgoing/) I always
get 'billsec=12' in the cdr, both mysql
I had this problem too, to fix it I had to add
[general]
vmexten=12345 ; extension to match in extensions.conf, default
'asterisk'
fromdomain=192.168.1.2 ;ip address of server, without this the voicemail
address asterisk passed to the phone was '12345@' and no domain part so
the phone just
Yeah, http://www.voiptalk.org with one registration you can receive as
many numbers as you like.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of scott
Sent: 24 January 2006 09:14
To: asterisk-users@lists.digium.com
Subject:
Hmm see if this works,
Extensions.conf
[ring-30]
exten = s,1,ringing();
exten = s,2,wait(30);
exten = s,3,hangup();
...
then in your Dial(SIP/1SIP/2Local/[EMAIL PROTECTED]);
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian J.
As far as I know there will be no difference.
32bit runs natively on AMD64 chips.
The only advantage of 64bit is the extra address space and huge integers
:)
But I could be wrong, iv not done any benchmarking myself just what i
have read on the net.
-Original Message-
From: [EMAIL
I assume the bluetooth connects as a hands free device and not a data
cable?
Iv not seen any mobile that will pass voice down the data cable.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Facundo Ameal
Sent: 08 February 2006
It will probably be easier to write an AGI
script to do this, I cant think of anything in the dialplan to do this.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arne Morten Johansen
Sent: 08 February 2006 13:38
To: Asterisk Users Mailing
to set a variable
in asterisk to TRUE or FALSE based on the result of the PHP-script.
Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Morgan Gilroy
Sendt: 8. februar 2006 15:28
Til: Asterisk Users Mailing List -
Non-Commercial Discussion
Emne: RE: [Asterisk-Users] GotoIf
I don't think asterisk can do this without hacking it.
You will need either 2 asterisk process one on each ip or have SER
listening on the second ip that just dumb forwards all packets back and
forth.
So to bill the second account you would dial at your second ip address,
ser will then forward
Yeah I do this,
create 2 ssh
sessions to the same box,
on the first session
do `script -f /tmp/astcli`
`asterisk r`
(and whatever other options you need
on the second session
`tail f /tmp/astcli | grep -i bob` (on the grep you
may have to ignore control chars
1
SUMMARY:Accepted: [asterisk-users] [Invitation] VoIP Users Conference @ Fri
Mar 712:00 - 13:00 ()
UID:[EMAIL PROTECTED]
ATTENDEE;ROLE=REQ-PARTICIPANT;PARTSTAT=ACCEPTED;RSVP=TRUE;CN=Morgan Gilroy
:MAILTO:[EMAIL PROTECTED]
ORGANIZER:MAILTO:asterisk-users@lists.digium.com
LOCATION:http
Hi,
Does anyone know how to get the codec that was negotiated for a call
after a dial? I want to log them into CDR but can't find any way to do
it without hacking the code.
It would be good if I could get it in an asterisk variable I can log off
seperatly.
Thanks!
directory
for
an example of how to do this.
-and-
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
Morgan Gilroy wrote:
Hi,
Does anyone know how to get the codec that was negotiated for a call
after a dial? I want to log them into CDR but can't find any way to do
it without
Of James
FitzGibbon
Sent: 11 May 2007 15:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Log CODECS in CDR's
On 5/11/07, Morgan Gilroy [EMAIL PROTECTED] wrote:
At the moment to find the codecs used I have to look though the
sip
trace
In your dial lines you have an extrac comma (,)
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
should be
exten = _9xxx,1,Dial(${OUTBOUND}/${EXTEN:1})
or
exten = _9xxx,1,Dial,${OUTBOUND}/${EXTEN:1}
From: [EMAIL PROTECTED]
[mailto:[EMAIL
problem or load problem, but during
these times conferencing etc works ok and there is no appreciable load
on the server or network.
Anyone have any ideas?
Thanks.
Morgan Gilroy.
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.
Morgan Gilroy
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Also check out http://www.bicom.us pretty expensive but if that's your
thing :)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ronald Hartmann
Sent: 27 September 2005 16:47
To: 'Asterisk Users Mailing List - Non-Commercial
Of Morgan
Gilroy
Sent: Tuesday, September 27, 2005 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Software only Asterisk PBX (commercial)
Also check out http://www.bicom.us pretty expensive but if that's
your
thing :)
-Original
the software and support to several guys
who
paid for the software.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Morgan
Gilroy
Sent: Wednesday, September 28, 2005 8:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'
From an AGI script so people can dial #* to hang up (and other things) but
when using AGI using 'EXEC DIAL IAX2/bla/1234567|50|tT' it dials ok but
nothing happens when they dial #, is there something special I need to do to
escape
, Morgan Gilroy wrote:
Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'
From an AGI script so people can dial #* to hang up (and other things) but
when using AGI using 'EXEC DIAL IAX2/bla/1234567|50|tT' it dials ok but
nothing happens when they dial #, is there something special I
Hi I have been trying to get SNOM (320,360) and hotdesking working with
asterisk.
I can get it working fine with SER but it fails with asterisk unless I
have no SIP password/secret in sip.conf
This is how it works with SER,
1. reset phone (removes accounts)
2. phone prompts for username and sip
Ah cool, thanks ill look at it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Maik Schmitt
Sent: 09 January 2006 11:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SNOM
Hi, I have now managed to get it working with asterisk 1.0.10 I had to
modify the patch http://bugs.digium.com/bug_view_page.php?bug_id=6035 as
its for the latest version of asterisk but it works very well now.
Thanks for the pointer.
-Original Message-
From: [EMAIL PROTECTED]
Can anyone give me some pointers on the following, in our setup (ast
1.6.3) we use international carriers to terminate calls for a
callingcard system, we have an issue where there can be a very long
delay after dialing but before the far end begins to ring.
I would like to play a tone every
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