Hi all,
Perhaps a no-brainer, but i think i am making my dialplan on my proxy
too complicated.
Normally, what you find in the examples is that you have to dial a
specific number, other 9 or 0 for an external line.
What i want to do is this:
If you pre-pend a number with something like * then
Hi
We're using asterisk 1.8.3.2 with the patch from issue 18818
Were finding a high incidence of channels staying open after the call
has finished when the call has been picked up using *8
I know there has been an issue with parking calls in 1.8, could this be
related?
our features.conf has
Please ignore, I should have looked at issues.asterisk.org first
https://issues.asterisk.org/view.php?id=18654
Apologies
Ish
On Fri, 2011-07-15 at 09:03 +0100, Ishfaq Malik wrote:
Hi
We're using asterisk 1.8.3.2 with the patch from issue 18818
Were finding a high incidence of channels
Most *8 pickup issues have been fixed in trunk. May have made it into 1.8.5,
I'm not sure.
https://issues.asterisk.org/view.php?id=18654 and others search mantis for
closed issues and 'pickup'.
Or newer https://issues.asterisk.org/jira/secure/Dashboard.jspa
Alec Davis
-Original
Beat me to it.
There are other commits that follow up from 18654 that may also help.
Check the blame's for changes to apps/app_directed_pickup.c and
main/features.c
Alec
-Original Message-
From: asterisk-users-boun...@lists.digium.com
sir,
is there any idea for this whenever 667and668 extension will dial isd call
before connect agent will dial password like ..
Best Regards,
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75
I'd be grateful if anyone here could comment knowledgeably on an idea
that I have had, as to whether it should be possible or not.
Consider two Asterisk boxes, each with one or more E1s on EuroISDN.
Each box has a different telephone number that hunts across all its
E1 channels. In addition there
Tony Mountifield wrote:
I don't want just to relay the call through to the second box using
IAX or SIP or an additional PSTN channel. What I would like to do is
to redirect the call in the PSTN so that it ends up connected only to
the second box.
If I recall correctly, it's only possible if
Hi Jan,
With PCI-passthrough or pure-SIP?
Mike
We're running Asterisk in a domU Xen VM.
Works great, including conferences, but we can predict the availability of
hardware resources.
--
_
-- Bandwidth and Colocation
We are building an app that will initiate outbound calls using .call files, and
each call can be a different duration (eg: 1min to 5min). These calls will go
through an Asterisk service with other calls/apps running.
I need to control the MAX number of channels in use so I don't overload this
Hy,
I try to change the linear parameter to rrobin with memory and nothing
has changed.
Here is the asterisk log :
/[Jul 15 14:30:05] VERBOSE[25232] app_dial.c: -- Called 5030
[Jul 15 14:30:05] VERBOSE[25230] app_queue.c: --
Local/5030@from-queue-ba73;1 is ringing
[Jul 15 14:30:05]
Tony Mountifield wrote:
I don't want just to relay the call through to the second box using
IAX or SIP or an additional PSTN channel. What I would like to do is
to redirect the call in the PSTN so that it ends up connected only to
the second box.
Doug wrote:
If I recall correctly, it's
I would suggest Two B-Channel Transfer (TBCT), transferring to a
unique
number (received as DNIS by the other server) that would identify the
call
as transferred from the first server and, perhaps, the reason for the
transfer.
It looks like TBCT may not have been implemented in Asterisk
In article 296076780.5348.1310743930593.JavaMail.root@zimbra,
Richard Mudgett rmudg...@digium.com wrote:
I would suggest Two B-Channel Transfer (TBCT), transferring to a
unique
number (received as DNIS by the other server) that would identify the
call
as transferred from the first server
I need to keep out all connection from 5 countries, which originate
most of the Denial of Service attacks. The entries are
around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter
way to do this by using User Tables in iptables, that will keep the
speed equal to LOG(x). I already tried
On 07/15/2011 12:47 PM, CDR wrote:
I need to keep out all connection from 5 countries, which originate
most of the Denial of Service attacks. The entries are around 9000 if
used as xx.xx.0.0/16. I heard that there is a smarter way to do this
by using User Tables in iptables, that will keep the
In article 296076780.5348.1310743930593.JavaMail.root@zimbra,
Richard Mudgett rmudg...@digium.com wrote:
I would suggest Two B-Channel Transfer (TBCT), transferring to a
unique
number (received as DNIS by the other server) that would identify
the
call
as transferred from the
On Fri, Jul 15, 2011 at 12:47 PM, CDR vene...@gmail.com wrote:
I need to keep out all connection from 5 countries, which originate
most of the Denial of Service attacks. The entries are
around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter
way to do this by using User Tables in
On Fri, Jul 15, 2011 at 12:47 PM, CDR vene...@gmail.com wrote:
I need to keep out all connection from 5 countries, which originate
most of the Denial of Service attacks. The entries are
around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter
way to do this by using User Tables in
Have just tried to test an upgrade to 1.8.5 and when making an outbound call I
get:
[Jul 15 18:48:52] WARNING[21038]: pbx.c:4071 pbx_extension_helper: No
application 'Macro' for extension (context, XX, 1)
I back leveled to 1.8.3 and that works fine. What am I missing as app_macro has
I need to keep out all connection from 5 countries, which originate
most of the Denial of Service attacks. The entries are around 9000 if
used as xx.xx.0.0/16. I heard that there is a smarter way to do this
by using User Tables in iptables, that will keep the speed equal to
LOG(x). I
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I missing as app_macro has
been installed okay?
Macro was depreciated in 1.6 and most likely removed in 1.8.5
I'd check the upgrade.txt to see if there is an entry about it, from the
wiki
Macro Implementation.
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I missing as
app_macro has been installed okay?
Macro was depreciated in 1.6 and most likely removed in 1.8.5
Removed, no. However in future version of Asterisk it will not be
I'm still using macro with Asterisk 1.8.5.0
On Fri, Jul 15, 2011 at 3:17 PM, Paul Belanger pabelan...@digium.com wrote:
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I missing as
app_macro has been installed okay?
Macro
Hello,
I've just upgrade from 1.4.2.20 to 1.8.3.1 and some kind of endpoint
aren't able to register. Message is:
[Jul 16 01:26:15] NOTICE[25443]: chan_sip.c:23511
handle_request_register: Registration from 'sip:user637801' failed for
'X.X.X.X:5060' - No matching peer found
sip.conf
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