ip entry, for OPTIONS to return a 200 instead of 404.
It doesn't matter what the 's' extension does, so it can just call Hangup.
Cheers
Tony
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__
In article <88f96e46-e6bb-a7ef-bebb-5588ef6cd...@gmx.ch>,
Fourhundred Thecat <400the...@gmx.ch> wrote:
> > On 2020-06-02 17:48, Tony Mountifield wrote:
> > In article <94191802-6c9c-bdab-615b-001786a2a...@gmx.ch>,
> > Fourhundred Thecat <400the...@gmx.
m I missing something?
I agree with you that it is strange the two logging types are different.
But someone with a different opinion than yours might well say "Why did
they decide to omit the line number and function from the file logging?
It's very useful information!"
The be
line you quoted:
same => n,NoOp(PAI=${PAI})
Then turn on verbose logging and try the call. Look at the logged
NoOp line and see if it contains just the 'John' or the whole value
'"John Doe" '
If it contains the whole value, then the problem is in the AGI library
t need to speficy the __ when reading the variable, just use ${PAI}
as before.
See https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance
Cheers
Tony
--
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Work: t...@softins.co.uk - http://www.softins.co.uk
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--
_
2. Macro(record)
[extensions.conf:881]
3. Set(CALLERID(num)=044111)
[extensions.conf:882]
-= 2 extensions (5 priorities) in 1 context. =-
Notice that the "n" converted to "4&quo
ing to
Asterisk or from Asterisk.
Do you have internal_timing set in asterisk.conf?
What timing module are you using?
Does it always happen, or just sometimes?
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.
oad. Nevertheless, we will try what you just posted.
Even if you put "exit 0" at the top of the script, the perl interpreter will
still need to compile the whole script (and any modules it uses) before it
executes the "exit 0".
Try commenting out or removing the rest of the
to
handle "qualify".
So in your [trunkinbound] context, just add a line like this:
exten => s,1,Hangup
And leave everything else in that context unchanged.
Cheers
Tony
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@
Who is the male voice artist who recorded the en_GB sounds for Asterisk?
Would be useful to know in case of the need to get additional matching
sounds recorded.
Cheers
Tony
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In article ,
Jean-Denis Girard wrote:
> Le 20/07/2019 à 12:21, Tony Mountifield a écrit :
> > What is the bug with channel variables? Do you have a fix for it?
>
> Channels variables caused an error, my fix is in aioswagger11/client.py
> (line 80)Â :
>
In article <301a2e78-d490-3805-e30f-41b668aac...@sysnux.pf>,
Jean-Denis Girard wrote:
>
> Hi Tony,
>
> Le 20/07/2019 à 06:29, Tony Mountifield a écrit :
> > Are there any other languages/libraries I should be considering?
>
> Same here, after years of AGI /
languages/libraries I should be considering?
Thanks for any advice!
Cheers
Tony
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_
-- Bandwidth
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/displa
ording on Asterisk 13?
>
> (This obviously is fatal anyway as I got lots of phones on which I want to
> playback recordings and editing /etc/hosts for each phone is impossible if
> two phones want to listen to different recordings at the same time-
> /etc/hosts can only contain one &
Dial() to propagate the answer, busy or other failure from
the destination channel back to the originating channel.
Is it possible that the setup part of the call (between initiation and answer)
is recorded in a separate CDR?
Cheers
Tony
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Work: t...@softins.co.uk - http://www.
binary, recompile, then compare the first binary with
the recompiled one? At the simplest level use "cmp -l". Or maybe convert
each binary to a hexdump with "hexdump -C", and then use diff or vimdiff
to compare them.
Cheers
Tony
-
ay to tell AMI that I don't want it to log login attempts - or,
> to put it another way, is there any way to tell the logger module to ignore
> AMI?
Look in /etc/asterisk/manager.conf for the option "displayconnects = yes/no".
It can be set globally in [general] or indiv
In article <7d8dc02f-0fce-4d47-72d9-604994c33...@palosanto.com>,
Alex VillacÃÂs Lasso wrote:
> El 29/05/18 a las 05:24, Tony Mountifield escribió:
> > In article <3a005ff6-19a4-215b-4751-bee616ec7...@palosanto.com>,
> > Alex VillacÃÂÃÂs Lasso wrote:
> >&g
nt ast_do_masquerade(struct ast_channel
} exchange;
struct ast_channel *clonechan, *chans[2];
struct ast_channel *bridged;
+#ifdef I_THINK_THIS_IS_WRONG /* Tony Mountifield, 2018-03-29. Removing this
code fixes lost CDRs with masquerade */
struct ast_cdr *cdr;
+#endi
In article <20180404133024.kpidrkuiyjoqd...@xorcom.com>,
Tzafrir Cohen wrote:
> On Wed, Apr 04, 2018 at 11:28:33AM +0000, Tony Mountifield wrote:
> > In article
> > ,
> > Richard Mudgett wrote:
> > >
> > > The libpri makefile doesn't install
In article ,
Tony Mountifield wrote:
> In article
> ,
> Matt Fredrickson wrote:
> > On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield wrote:
> > > In article
> > > ,
> > > Matt Fredrickson wrote:
> > >> That does seem quite odd. If I rem
when dahdi_cfg couldn't find libtonezone).
Would there be any subtle issues with the 64-bit libraries being loaded
from /usr/lib instead of /usr/lib64?
Should Asterisk and DAHDI builds also be updated to use /usr/lib64 when
building on a 64-bit OS? Or the build instru
In article ,
Matt Fredrickson wrote:
> On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield wrote:
> > In article
> > ,
> > Matt Fredrickson wrote:
> >> That does seem quite odd. If I remember right, those messages would
> >> come up if it looked like the o
degrade to half-duplex trying to talk
to full-duplex, resulting in lots of collisions and packet loss when there
is any kind of significant traffic.
Your description would be consistent with the firewall introducing lots of
LAN collisions when busy, in the central gigabit switch, even if the VoI
In article ,
Matt Fredrickson wrote:
> On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield wrote:
> > I have some more investigation to do on this, but I wanted to see if anyone
> > here had any insight into the issue I've run into.
> >
> > The hardware is a HP DL360
oes anyone have any clues why there would be a difference
in PRI behaviour between 32-bit and 64-bit builds? Has anyone else run into
anything similar?
Cheers
Tony
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o send to an unreachable peer), that may set
DIALSTATUS without setting HANGUPCAUSE.
So HANGUPCAUSE should be considered as extra detail, rather than a replacement
or alternative to DIALSTATUS.
Cheers
Tony
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mount
l(SIP/officephone,120,m)
>
> [secondline]
> exten => 22,1,Dial(SIP/livingroomphone,120,m)
>
> [thirdline]
> exten => 33,1,Dial(SIP/bedroomphone,120,m)
But because you have all three of your trunk peers pointing to the
same context, you don't necess
xample) is not used to select or match the inbound SIP
peer.
When the call comes in from sipgate, it probably doesn't have a fromuser.
The fromuser can be used to select the peer based on matching the [string]
that names the peer.
Otherwise, when Asterisk is looking for a matching peer sect
files
available in all the possible native formats. Then Asterisk can use the
appropriate
one for the channel without transcoding.
On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729.
They will also sound better than transcoding from the gsm versions.
Cheers
T
for console, then AFAIK, you won't
see verbose messages however many -v options you give it!
Cheers
Tony
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__
tart it in /etc/rc.d/rc.local for want of anywhere better.
Being in /var/tmp, cron.daily/tmpwatch deletes files older than 30 days.
I could just have easily put them somewhere else and used the -W option
to tcpdump to remove old files on a rolling basis.
Cheers
Tony
--
Tony Mountifi
hat extension has been answered and it is safe to play the
message.
That is why I added the comment "how long???", as it is just a guess.
Cheers
Tony
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--
t have the "app_mysql" module.
If you compiled asterisk yourself, you need to go into menuselect and make
sure app_mysql is selected, and then recompile.
You will probably find app_mysql under "Addons".
If it does not let you select app_mysql, you will need to install the my
om the same kernel too (make sure
you rebooted after any update of the kernel).
The version of kernel-devel you have installed is: 3.10.0-327.36.3
The version of kernel-devel the makefile wants is: 3.10.0-327
The difference is significant, and suggests that you are actually
running an older ke
only way to do that is the
> gH workaround above.
>
> If I'm not missing a trick here and there's no better way to do those
> to things, is there any way to force Asterisk to NOT "optimize" those
> channels?
Yes, append /n to the local channel:
sam
In article ,
Jonathan H wrote:
> On 13 October 2016 at 13:18, Tony Mountifield wrote:
>
> > exten => _X,1,NoOp(Matching single digit)
> > exten => _X.,1,NoOp(Matching multiple digits)
> > exten => _X!,2,SayNumber(${EXTEN})
> > exten => _X!,3,Etc..
&
bout priority 1, so that's the only priority you need to double.
You should be able to use ! safely in priority 2 upwards:
exten => _X,1,NoOp(Matching single digit)
exten => _X.,1,NoOp(Matching multiple digits)
exten => _X!,2,SayNumber(${EXTEN})
exten => _X!,3,Etc..
Disclaimer: I ha
ule am I missing?
The ExecIf command is provided in the module app_exec, which is usually
located at /usr/lib/asterisk/modules/app_exec.so
Maybe you had turned off app_exec in the menuconfigi when building, or maybe
your
modules.conf has a
27;t get a number)
Maybe that's what is happening in your case, so try adding an "s" extension.
Hope this helps,
Tony
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Work: t...@softins.co.uk - http://www.softins.co.uk
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_
ll Waiting enabled, and the waiting call
has been ignored by the recipient until it times out.
See https://en.wikipedia.org/wiki/Call_waiting and try *#43# on the mobile
in question to check whether call waiting is active. Use #43# to try
deactivating it and see if that helps.
Cheers
Tony
-
In article ,
Jerry Geis wrote:
>
> > You need to have installed DAHDI before compiling libpri or asterisk
>
> Thanks Tony,
>
> in fact I have compiled DAHDI complete 2.11.0+2.11.0 - It did not show any
> errors.
> I did make; make install on the DAHDI source
>
directory.
You need to have installed DAHDI before compiling libpri or asterisk.
Cheers
Tony
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
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___
t doesn't show
any Context, Extension and Priority. Where is the channel supposed to go
once the call to SIP/430 is answered?
3. The Asterisk "full" log, with at least verbose level 3, encompassing
your attempt.
4. Anything else that you yourself would need to look at to debu
id "first time" trying this.
>
> Any thoughts?
Not really. Very little info to go on so far. You need to give us
more detail of what you are doing with AGI and AMI.
Cheers
Tony
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mo
0)
> ConfBridge(to take this user into above conference)
>
> But the SIP/100 and SIP/101 calls do not take place until a second delay.
>
> Why are the SIP/100&SIP/101 calls delayed during the Wait(10) ?
Are you saying that this worked in earlier versions but you started to
g
to vary between reboots or rebuilds?
3. Is there a way I can make the order predictable and fixed? modules.conf?
4. Or alternatively, a way I can make MRTG find the correct index dynamically?
Thanks for any advice,
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play
you can describe how you have set up your FastAGI server, and how it
invokes DatabaseQuery.agi, that would help us to help you!
Cheers
Tony
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
--
__
ree will be served.
I have this:
## incl/excl subtree mask
view allincluded .1 80
##context sec.model sec.level prefix read write notif
access MyROGroup "" any noauth exact all
In article <20151125133008.6369360.14455.17...@gmail.com>,
Israel Gottlieb wrote:
> Try putting progress instead of answer
Yes, I tried Progress already, and it didn't help. But thanks for
the suggestion!
Tony
> I have a puzzling situation, and would be grateful for any insig
els wrong that I should.
The siptrunk entry contains canreinvite=no and directmedia=no.
The version of Asterisk on these boxes is 10.5.1, if that's relevant.
Thanks for any insight!
Cheers
Tony
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@m
hen
> >finished with).
>
> Hi Tony
>
> Thanks for replying.
>
> I suspected something like that, though repeatedly running
>
> lsof | wc -l
>
> Always stays quite low - 100 000 open files, which is still 8 times less
> than the system maximum as confirmed b
uite possible, even probable, that such a leak
has been found and fixed, even in the 1.8 series. 1.8.11.0 is rather old -
the latest is 1.8.32.3, so it would be best to update to that version and
see if the problem persists.
Cheers
Tony
--
On Tue, Jun 30, 2015 at 2:15 PM, Duncan Turnbull
wrote:
>
> Hi Tony
>
> I'm not familiar with the card you but 120 ohm is usually twisted pair,
> and 75 ohm is coax (usually). If it is changeable its usually done with
> jumpers on the card.
>
The new Digium cards hav
modem that
somehow doesn't work with our cards but am struggling to find the technical
reasons (and possible fixes) to support that hypothesis.
On Tue, Jun 30, 2015 at 1:43 PM, Dale Noll wrote:
>
> On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule wrote:
>
>> Hello,
>>
>&
3010 desktop with
another TE235 card in it but same results! I can confirm that the cards are
ok!
On Tue, Jun 30, 2015 at 12:36 PM, David Duffett wrote:
> What response do you get to *CLI> pri show spans ?
>
> On 30 June 2015 at 09:34, Tony Kasule wrote:
>
>> Hello,
>>
Hello,
Anyone to help me with this issue? It has never worked :(
On Wed, May 20, 2015 at 11:34 AM, Tony Kasule wrote:
> Hello users,
>
> I have a Digium Te235 and asterisk 13 which have worked well with 1
> carrier but we have failed to add a 2nd carrier. The second telco bring
ifically trying to do is to determine historically the
usage of the G.729 licences installed in a system, but an answer to
the more general question would be useful.
Thanks
Tony
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Play: t...@mountifield.org - http://tony.
Hello users,
I have a Digium Te235 and asterisk 13 which have worked well with 1
carrier but we have failed to add a 2nd carrier. The second telco brings
their E1 line over finer, terminated in a RAD modem and they give me
ethernet to the E1 card. It's the first time i am having install such a
so
In article ,
Tony Mountifield wrote:
> I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that
> is behind a network device to which I don't have ready access, which is
> performing NAT with possibly some kind of SIP ALG, and an Asterisk 11
> system on a p
#x27;ve changed the address of the public endpoint):
Mar 30 10:20:20 VERBOSE[5811] logger.c: Retransmitting #5 (no NAT) to
11.111.11.111:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
11.111.11.111:5060;branch=z9hG4bK6bee4b53;received=11.111.11.111
From: "Tony Mountifield" ;tag=as4ab948f7
To
In article ,
Grant Bagdasarian wrote:
>
> Is it possible to log the raw signaling of Dahdi channels to a log file?
Try googling for: dahdi pcap
It should be possible to log to a pcap file that you can later examine
using Wireshark. I haven't yet tried doing so.
Cheers
To
this system to change ATTRS back to SYSFS? I notice
that it was changed to ATTRS in DAHDI 2.9.
2. Should /dev/dahdi/devices contain just @Board, or also 1/ and 2/?
3. Are the /dev/dahdi/devices entries used by Asterisk or anything else?
Thanks for any advice!
Cheers
Tony
--
Tony Mountifield
Wo
the instructions here:
http://www.builddesigncreate.com/index.cgi?mode=webpage_list&pageid=2011080413332724848
If that prevents the problem, the next step would be to determine why
pre-linking causes the problem, although I&#x
www.asterisk.org appear no longer to exist.
I already have the core- (good quality) and extra- (poor quality) sets
of standard prompts.
On a related note, the extra- set appears to have been converted from
the old GSM format. Are there any plans to have them re-recorded in
good quality?
Cheers
Tony
ot just to give up on it. There's no reason it can't be made
to work correctly, and it enables RTP ports to be opened and closed
as required, instead of having a complete range permanently open.
Such a pity WatchGuard is closed-source.
Cheers
Tony
> -Original Mess
In article
,
Ishfaq Malik wrote:
> On 22 April 2014 16:24, Tony Mountifield wrote:
>
> > Has anyone here used Asterisk inside a WatchGuard firewall, talking via
> > the WatchGuard SIP Application Layer Gateway to an outside SIP service?
> >
> > I have a customer
rewrites it. However, either they or WatchGuard will not accept there
is a bug, despite my very detailed description of it.
So if anyone else has any experience of using this product, I'd be very
interested to hear from you. Thanks!
Tony
--
Tony Mountifield
Work: t...@softins.co.uk -
In article ,
Tony Mountifield wrote:
> In article
> ,
> Ishfaq Malik wrote:
> > On 11 April 2014 11:34, Tony Mountifield wrote:
> >
> > > Are they any gotchas to be aware of in getting Asterisk and Lync 2013
> > > talking to each other
In article ,
Ishfaq Malik wrote:
> On 11 April 2014 11:34, Tony Mountifield wrote:
>
> > Are they any gotchas to be aware of in getting Asterisk and Lync 2013
> > talking to each other using SIP? Or is Lync a pretty standard
> > implementation
> > of SIP?
>
&g
Are they any gotchas to be aware of in getting Asterisk and Lync 2013
talking to each other using SIP? Or is Lync a pretty standard implementation
of SIP?
Cheers
Tony
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eudo channel to give it data to mix.
And yes, two pseudos per meetme - one for recording from and one for
playing announcements into the conference.
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.s
In article ,
Paul Belanger wrote:
> On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield wrote:
> > I haven't been able to find the answer online, and am not currently
> > able to conduct an experiment to find the answer...
> >
> > I understand that in a SIP call w
r a legacy system),
and also whether it is any different on later versions.
Thanks,
Tony
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
--
_
-- Ban
In article <9519872.915.1383925949785.JavaMail.myoung@myoung-laptop>,
Michael L. Young wrote:
> > From: "Tony Mountifield"
> > To: asterisk-users@lists.digium.com
> > Sent: Friday, November 8, 2013 10:39:25 AM
> > Subject: [asterisk-users] 11.5
rying?
I checked sip.conf, and registerattempts was left unset (defaults to 0=forever).
I couldn't see any other relevant settings.
Cheers
Tony
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--
__
problem is a misconfiguration at the remote
end. I had that once, where the PBX to which Asterisk was talking had had
its channel numbers misconfigured, resulting in a similar problem to what
you have described.
What happens if you swap the cables over between the two E1 ports on the card?
Does t
Card 0 Span 2"
> group=0,12
> context=from-pstn
> switchtype = qsig
> signalling = pri_net
> channel => 32-46,48-62
> context = default
> group = 63
>
> could you please help me
What are the contents of /etc/dahdi/syste
rical problem that
has largely gone away nowadays. You will get better quality with G.711 at least.
Cheers
Tony
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--
_
miator (I haven't checked), but it certainly
sends \r\n to terminate lines that it outputs. IMHO, it's good to adhere
to the same convention in both directions.
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@m
$resp = <$s>;
#print $resp,$line;
if ($resp =~ /Status: (.*)\n/) {
$status = $1;
} else {
$status = 'Unknown';
}
$spans[$span-1] = "Span $span status = $status\n"
y wrong too, and should be 1-1)
It also means that you should allow at least twice as many ports as the
number of simultaneous calls you want to handle.
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t.
it.
4. Else if it's a new ssh connection, accept it.
5. Otherwise reject it.
Nothing in there about accepting UDP, which is why you needed the extra
rule to accept the IAX port.
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
In article ,
Sean Darcy wrote:
> On 09/07/2013 10:33 AM, Tony Mountifield wrote:
> > In article <522a934d.8010...@gmail.com>,
> > Sean Darcy wrote:
> >> On 09/06/2013 07:08 PM, Steve Edwards wrote:
> >>> On Fri, 6 Sep 2013, Sean Darcy wrote:
> >&
7;t asterisk seeing/acting upon the registration request?
> Wireshark finds the packet to 4569, so it's not a firewall problem.
Are you sure about that? I have found in the past that tcpdump sees inbound
packets before they get to the iptables filter.
What happens if you do:
iptables -I INPU
In article ,
Tony Mountifield wrote:
> I have a system running CentOS 5.9 and DAHDI 2.6.2 with a 2-port E1 card
> using the wct4xxp driver (also using Asterisk 11.5.0, but that isn't
> relevant to the question).
>
> With DAHDI and Asterisk started, the system appears to run n
uinfo
reports 8 CPUs, but I don't know whether that is just coincidence. The CPU
is a X3450 with four cores and HT enabled.
Any thoughts would be gratefully received!
Cheers
Tony
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - ht
In article
<616B4ECE1290D441AD56124FEBB03D08171492B22C@mailserver2007.nyigc.globe>,
Eric Wieling wrote:
> This is the standard way we set up our servers. There is nothing special
> about it. Just make sure you disable direct media.
Thanks, that's reassuring. Appreciate th
In article <51f925f2.1040...@dns99.co.uk>,
Gareth Blades wrote:
> On 31/07/13 15:32, Tony Mountifield wrote:
> > Most of my experience until recently has been in Asterisk 1.2, and I am
> > just starting to make use of Asterisk 11 for new systems.
> >
> > I hav
test system,
I wondered if anyone here has made such a setup, and whether there are
any issues with getting SDP contents and media routing correct?
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifiel
quot;]]?notfromlocal:)
>
> But I am getting a message say there is no variable to check. So what
> I have done that is wrong?
Is that step split into three lines in your dialplan? I think you might
need to put it all on a single line.
Cheers
Tony
--
Tony Mountifield
Work: t...@
hat presumably provides the drivers with the ability to capture to
pcap files, and a tool to control it.
Presumably a recent version of Wiresharl will then be able to interpret
the captured files.
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http
ing any pri card thats supported on Asterisk.
And you may need to make an E1 crossover cable. These are different from
Ethernet crossover cables. You need to cross pair 1-2 with pair 4-5.
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.soft
anks. But I found the right syntax now:
> Exec Set CALLERID(num-pres)=prohib
>
> This AGI-Command leads into "200 OK" and I can verify, that outgoing
> calls (SIP and DAHDI) are anonymous.
Yes, that will work, but it is executing the Set() dialplan application.
AGI has
In article <20121108092952.78cb6...@ws78.int.tlc>,
Chad Wallace wrote:
> On Thu, 8 Nov 2012 16:44:32 + (UTC)
> t...@softins.co.uk (Tony Mountifield) wrote:
>
> > Here is a simplified example:
> >
> > [test]
> > exten => _X.,1,Dial(Local/${EXTEN}@o
hile it was still ringing.
So I understand the reasons for the above behaviours, but my question
is: How can I propagate the NOANSWER status upwards from the inner Dial,
so that the Local channel also returns NOANSWER?
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.soft
h Dial(), Goto(), etc), at an extension that exists. If it
doesn't exist, the context cannot be entered.
The 'i' extension is only used when already in a context, and is mainly
for catching unmatched extensions dialled within a Background or WaitExten.
See http://ww
I could not
find comparable information on the Asterisk WIKI at https://wiki.asterisk.org).
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
--
__
volved. The problem is that the PSTN
will not drop the call when the called party on an analogue line hangs
up, until after a long timeout. There is usually no solution to this.
Cheers
Tony
> On Tue, Sep 18, 2012 at 2:03 PM, Tony Mountifield wrote:
>
> > In article <
> > caehs
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