compiling natively or cross compiling.
I am compiling natively.
Doug
On Tuesday, March 4, 2014 12:54 AM, Julian Lyndon-Smith aster...@dotr.com
wrote:
this is all very odd. I have been compiling on raspbian wheezy for a
few months now, and have never come across this error
-rw-r--r-- 1
://lists.digium.com/mailman/listinfo/asterisk-users
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don't care if it works on your machine! We are not shipping your machine!
The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
.
This would necessitate that the conversation is monitored and analysed
in realtime as we can't do it post-call ;)
Thanks
Julian
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your machine!”
The kangaroo dances: http://www.youtube.com
:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your
machine
)})},
${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})}
In the actual configuration file, the value in the Master.csv mapping
should be on a single line.
cdr_manager
On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote:
2013/3/29 Julian Lyndon-Smith aster...@dotr.com
check
is foo, the value set before entering the
hangup extension (see previous dialplan)).
To me, this is either a feature (you can't set CDR values in hangup exten)
or a bug.
How would you qualify this ?
2013/3/29 Julian Lyndon-Smith aster...@dotr.com
Ah, right. Have a look
that it's a big ask, and I will understand if you decline to do so,
but would really appreciate it if you could.
Many thanks for the insight, either way.
julian
On 14 February 2013 15:27, Jeremy Kister asterisk...@jeremykister.comwrote:
On 2/14/2013 1:20 AM, Julian Lyndon-Smith wrote
very polite *bump*
this is a real issue for us - anyone got _any_ clues or ideas ?
Thanks ;)
On 12 February 2013 14:29, Julian Lyndon-Smith aster...@dotr.com wrote:
Ever since we upgraded to asterisk 11 we have had audio problems with
our cisco 7940 phones.
The problems manifest
11 ?
Many thanks
Julian
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your machine!”
The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
/mailman/listinfo/asterisk-users
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your machine!”
The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your machine!”
The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
://lists.digium.com/mailman/listinfo/asterisk-users
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your machine!”
The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
you think it is necessary ? and the problem on LDAP is associate with
dahdi?
From: Julian Lyndon-Smith aster...@dotr.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, 8 November 2012, 9:40:50
/mailman/listinfo/asterisk-users
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your machine!”
The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
)
group_count(potential_${CONFNAME} )
however, when the call enters the conference, I need to unset the potential
calls groups
how do I do this ?
/me feels very very stupid
Julian
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your
for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works
wrote:
On 01/04/2012 04:07 PM, Julian Lyndon-Smith wrote:
this looks great - is there any chance of coverting the googletts.agi
to use flac as well ?
Julian
In googletts.agi we get the voice data from google in mp3 and we convert
it in a format that asterisk can read and playback (slin). If we
nope :(
On 4 January 2012 14:29, Lefteris Zafiris zaf@gmail.com wrote:
On 01/04/2012 04:24 PM, Julian Lyndon-Smith wrote:
the only reason is that I didn't want to have to install sox. Lazy.
that's all ;) Just another piece of software to find and install
running on amazon ec2
/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your machine!”
The kangaroo dances: http
minutes I
get the message
Unknown response to CalDAV calendar pug, request REPORT to
/calendar/myemail/events/: Could not read status line: connection
was closed by server
Has anyone managed to get this running ?
Julian
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works
'20-18295@127.0.0.1': while expecting '100' (index 1), received
'SIP/2.0 407 Proxy Authentication Required
What I am trying to figure out is how / why / what is different that
now asterisk requires proxy authentication for sipp, when it didn't
before.
Thanks
Julian
--
Julian Lyndon-Smith
://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t care if it works on your machine! We are not shipping your machine!”
The kangaroo
to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Julian Lyndon-Smith
IT Director, Dot R Limited
I don’t
Can someone confirm if the bug present in #18951 has been fixed in 1.6 or 1.8 ?
If not, then I am stuck on my current version of 1.4, and will not be
able to upgrade to either of those two versions, even for security
fixes.
Julian
On 19 April 2011 15:52, Paul Belanger pabelan...@digium.com
-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith
Sent: Tuesday, April 19, 2011 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Good by asterisk 1.4? Please
1.4 svn has a nasty bug in it at the moment. Would love to see that fixed ;)
https://issues.asterisk.org/view.php?id=18951
Julian
On 15 April 2011 14:22, Satish Patel satish...@hotmail.com wrote:
You know we don't have choise. I had remembered when we shifted 1.2 to first
release of 1.4 and
Anyone had any experience of using this phone with asterisk ? Trying
to find out if I can provision it using tftp / http
Thanks
Julian
--
Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker
--
_
-- Bandwidth
I think I've seen this where I am trying to start another instance of
asterisk using safe_asterisk, when I already have an instance running
Julian
On 16 October 2010 22:36, Dan Journo d...@keshercommunications.com wrote:
Hi,
Does anyone know where this is suddenly coming from?
--
the beginning.
Julian
On 29 August 2010 18:17, Tilghman Lesher tles...@digium.com wrote:
On Sunday 29 August 2010 03:32:07 Julian Lyndon-Smith wrote:
Still can't figure out how to fastforward / rewind the current file
being played.
core show application ControlPlayback
--
Tilghman Lesher
Edwards asterisk@sedwards.com wrote:
On Sat, 28 Aug 2010, Julian Lyndon-Smith wrote:
I want to be able to allow a caller to dial a ddi, system to verify
identity etc (this is all done)
I then want them to sit listening to music, until an event happens.
When this (external) event happens, I
, Julian Lyndon-Smith aster...@dotr.com
wrote:
Thanks Steve,
Not sure how this would allow the caller to ff / rw the file currently
being played - would that portion have to be written in the external
program ?
Are there any examples of how to use externalivr anywhere (I can't
find on google
I want to be able to allow a caller to dial a ddi, system to verify
identity etc (this is all done)
I then want them to sit listening to music, until an event happens.
When this (external) event happens, I want to play a certain file to
the caller, using playback (so that they have ff / rw etc),
...@venturevoip.com wrote:
On 22/08/10 10:38 AM, Julian Lyndon-Smith wrote:
Crap, sorry, meant to add that we are running 1.4 svn head
Julian
On 21 August 2010 23:38, Julian Lyndon-Smithaster...@dotr.com wrote:
We are having some strange issue where a call from asterisk dials a
mobile number. If the number
We are having some strange issue where a call from asterisk dials a
mobile number. If the number answers, we put the call through to an
agent SIP phone. All works fine.
If, however, the call goes straight through to the mobiles voicemail
service *and* the agent phone is a Cisco 79xx, then the
Crap, sorry, meant to add that we are running 1.4 svn head
Julian
On 21 August 2010 23:38, Julian Lyndon-Smith aster...@dotr.com wrote:
We are having some strange issue where a call from asterisk dials a
mobile number. If the number answers, we put the call through to an
agent SIP phone. All
It may be a bit outside Myth, but it's even further outside from Asterisk :)
Sorry, can't help.
Julian
On 22 August 2010 01:33, Michelle Dupuis mdup...@ocg.ca wrote:
I realize this is getting a bit outside myth...but hopefully someone can
offer some ideas...
I'm using the latest NVIDIA
You could always use the CURL function directly in the dialplan
Julian
On 10 August 2010 08:36, Tino t...@sparksupport.com wrote:
Hi Steve, thanks for your interest in this matter.
I will explain my requirement here.
In my asterisk server before an agent doing manual dial is allowed a
Anyone got a clue ? (he asks in desperation!)
Julian
On 9 July 2010 17:48, Julian Lyndon-Smith aster...@dotr.com wrote:
We are having a situation on our dialler here where our agents are
claiming that when they receive a call because it has been answered,
it seems as if the call had been
That looks like the option that will help a lot.
Thanks.
On 8 July 2010 23:21, Steve Edwards asterisk@sedwards.com wrote:
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
Lyndon-Smith
We have had 20 calls over the last month where the SIP channel has not
identified
We are having a situation on our dialler here where our agents are
claiming that when they receive a call because it has been answered,
it seems as if the call had been answered several seconds earlier -
IOW, they are hearing hello ? Hello ? and often hear the phone being
put down as an initial
We have had 20 calls over the last month where the SIP channel has not
identified that the person on the receiving end has hung up.
Is there a way of fixing this ?
TIA
Julian
--
_
-- Bandwidth and Colocation Provided by
We have a need for up to a dozen UK mobile numbers to be forwarded to
a UK landline. I know that I can just forward them, but was wondering
if anyone knew of any deals / contracts with a UK mobile operator that
would lessen the cost.
At the moment we are looking at going with Vodafone .
Thanks
Ted,
We've been using Asterisk in-house since 2005, with 100 people
connected. We are a call center, making approx 3000 inbound / outbound
calls per day 6 days a week. We have interfaces to 90 ISDN lines and
SIP providers. We use MOH, voicemail, queues etc etc, and record every
call.
Each agent
At the moment, we have a feature where if someone's sip extension is
called, we also make another call to their mobile. We use the c
option in the zap dialstring so that the user has to press # after
answering to confirm the call (this prevents things like the
answermachine grabbing the call if
There was a bug reported on this, I think ... yes #16581
Fixed in
r244070 | tilghman | 2010-02-01 11:46:32 -0600 (Mon, 01 Feb 2010)
Julian
On 17 February 2010 15:00, James Northcott / Chief Systems
ja...@chiefsystems.ca wrote:
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and
I have uploaded a patch for 1.4 and trunk that allows you to mute
either or both parts of a mixmonitor recording. I would appreciate it
if someone apart from me could test it and let me know how you get on.
Thanks!
Julian
https://issues.asterisk.org/view.php?id=16740
for PCI-DSS compliance we
A week or so ago, I explained that we need to blank our call
recording when some sensitive information like credit cards where
being discussed. With the lists help, I managed to find the pause/
unpause monitor commands. That works great. However (there is always
a however), what that now means is
During a telephone conversation with a customer, they sometimes give card
details over the phone. under the pci-dss regulations we are not allowed to
record the conversation where the details are being given. Is there a mute
command or pause that can be sent to MixMonitor ?
How has anyone else
Something similar along the lines of a previous email - has anyone
developed, or is using, something similar to this
http://www.veritape.com/wp-content/uploads/2009/11/veritape-call-tagging-module-description.pdf
Julian
--
_
--
+Monitor
Depending on your release, you can “pause” and “un-pause” monitoring.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Julian
Lyndon-Smith
*Sent:* Monday, January 25, 2010 8:22 AM
-info.org/wiki/view/Asterisk+cmd+Monitor
Depending on your release, you can “pause” and “un-pause” monitoring.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Julian
Lyndon-Smith
*Sent:* Monday
This is crazy. Something about writing to the list gives me ideas ;)
What I am looking for is
show manager command pausemonitor
;)
Thanks anyway, all.
Julian
2010/1/25 Julian Lyndon-Smith aster...@dotr.com
Oh, crap. the second I send, I realize I use features.conf, right
Anyone got any subjective (!) views on the merits of these two ranges ,
using asterisk 1.4 ?
I need to supply approx 30 handsets to a new client, with the senior
managers (6) having some slightly more managerial phones than the base
phones which will be used for one line only.
TIA
Julian
--
Has anyone managed to get these two phones to make a video call to each other ?
If so, care to share how the hell you managed ?
the GXV is at the latest firmware, and xlite the latest download
Asterisk 1.4 trunk
TIA
Julian
--
/14 SIP s...@arcdiv.com:
Julian Lyndon-Smith wrote:
Has anyone managed to get these two phones to make a video call to each
other ?
If so, care to share how the hell you managed ?
the GXV is at the latest firmware, and xlite the latest download
Asterisk 1.4 trunk
TIA
Julian
Yes. Have
We have a couple dozen Aastra 9133i phones in use - no problems
encountered, they worked well for us. However, these are now
discontinued. Does anyone have any views on the new product line up ,
or the Yealink phones ?
Julian
___
-- Bandwidth and
I'm just curious to know if anyone is using a usb 2.0 / ISDN30
(specifically EuroISDN) device. We are looking to purchase another pci
card, but was wondering if anyone has any horror / success stories to
share regarding a usb device.
TIA
Julian
___
--
-users-boun...@lists.digium.com] On Behalf Of Julian
Lyndon-Smith
Sent: Tuesday, December 08, 2009 8:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Interesting problem with IP's
Have a trunk 1.4 asterisk, running on centos on the lan at work.
A long
Have a trunk 1.4 asterisk, running on centos on the lan at work.
A long story, but we had the entire work network on a public address
range (90.1.0.x), going to a firewall, then out to the net.
At home (192.168.1.x network) I have a router that connects to the
firewall via a vpn tunnel.
All was
phone_office=+fullinternationalnumber
email=y...@email.com
homepage=http://www.example.com
avatar=/var/lib/asterisk/images/skype100x100.jpg
Just a note the country code has to be lower case (i.e GR would not work).
Panos
On Sun, Dec 6, 2009 at 9:40 AM, Julian Lyndon-Smith aster...@dotr.com
Aha. That was it. Thanks.
I could not see that advice in the documentation. I may be blind, but
it may be helpful to include it somewhere.
Thanks again
Julian
2009/12/6 Kevin P. Fleming kpflem...@digium.com:
Julian Lyndon-Smith wrote:
That's my point - SFA comes with a g729 licence, so why
As I have no friends and no life I thought that I would set up my
asterisk server with Skype.
1) Paid the $, got the licence, built and installed
2) create a business skype account (called company foo)
3) created a member of the business called bar
4) updated the skype conf file
5) restarted
translation path from 0x100 (g729) to 0x8 (alaw)
on the console.
Fired up a sip client, made the same call, and all was ok.
Any clues ?
2009/12/5 Julian Lyndon-Smith aster...@dotr.com:
As I have no friends and no life I thought that I would set up my
asterisk server with Skype.
1) Paid the $, got
Ok. So I bought 2x skpye channels. Doesn't that mean I have 2xg729 as well ?
If so, why do I have the problem ? And would this affect local
channels as well ?
Julian
2009/12/6 Kevin P. Fleming kpflem...@digium.com:
Julian Lyndon-Smith wrote:
external = ddi = dial(skype)
and got a load
Just for some information really : How many of you use multiple sip lines on
a phone ?.
I'm sitting here looking at my 7960, with it's 6 lines. I've every only used
one line, and I was wondering if I was a weirdo ;)
The only time I've ever found a use was when I had two systems (production
and
We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a
hotdesk type system where anyone can log on to an extension - however what
I would love to do is relabel the phone with the current owner when this
logon happens. I know that I can change the sip.conf and phones tftp file,
I was thinking of putting a cisco router on the E1 line for my test
system, so I can have multiple test servers accessing the ISDN, rather
than a dedicated server and a TE410 card.
I *am* confused at all of the modules for the cisco :)
What would be the best router to use to connect 30 channels
:00 AM, Julian Lyndon-Smith aster...@dotr.com
wrote:
I was thinking of putting a cisco router on the E1 line for my test
system, so I can have multiple test servers accessing the ISDN, rather
than a dedicated server and a TE410 card.
I *am* confused at all of the modules for the cisco
Using 1.4 svn, I want to implent the busy application.
With the following dialplan:
[inboundqueue]
exten = _X.,1,Answer()
exten = _X.,n,Goto(dropcall,1)
...
exten = dropcall,1,Busy(10)
exten = dropcall,n,hangup()
If I call any number in the inboundqueue, I get the following:
[Oct 1
We've been having a strange problem all day where when making outbound
calls, all we get is static on the far end (i.e we can hear, they
can't).
We've restarted asterisk a couple of times to no avail. It now
transpires that it is only mobile numbers that are affected (not all
mobile networks, not
getting some errors on my test system. this is 1.4 (Asterisk
SVN-branch-1.4-r214194) with a 4 port T412p card.
Three of the ports are connected: Span 1 to the PSTN on a 10 channel
bearer line, ports 2 and 3 are cross-overed (!) to each other. Port 4
is not plugged in. This has been working fine
I'm trying to figure out the maximum length of a cisco 7960 password
in the SIPmac.cfg file. An Aastra9133i can take at least a
36-character password, but the cisco craps out (can't authenticate)
In order to stop me from doing a brute-force test, does anyone know
the password lengths of
Aastra
Nope - but you are also running on an unsupported version of asterisk,
so I am not surprised. From the readme:
===[ Installation Overview ]===
It is required that the proper version of Asterisk is installed prior to
installing Skype For Asterisk.
What is interesting is that there is no mention of the software used -
if it is asterisk, he would need to make the code available, no ?
Julian
2009/7/18 Alan Lord (News) alansli...@gmail.com:
On 18/07/09 00:35, Gavin Henry wrote:
This has to be an Asterisk based appliance no?
Another simple way is to add local/foo/n as the only agent on the
queue. In the dialplan for local/foo , interrogate a database for the
most appropriate agent and then call that agent's extension.
Julian
2009/7/17 Matt Florell astma...@gmail.com:
On 7/17/09, Alex Balashov
We use a queue so that we can have all the benefits of the queue
whilst finding an agent : music on hold, periodic announcements etc
etc.
You are right - with a little more effort we could probably remove the
need for the queue. But why would I do that if I can use the queue for
the bits I want
Um, I really don't know - we just use the periodic messages to play
the traditional Your call is important to use (whatever the
wording..)
Julian.
2009/7/17 Alex Balashov abalas...@evaristesys.com:
What value do the queue announcements (I am assuming these are pertaining
to expected hold
Heh. See my previous posts ;)
We use curl to grab the agent info from the application.
Julian
2009/7/17 Leif Madsen leif.mad...@asteriskdocs.org:
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
We are trying to implement skill based routing for agents in a support
centre
Um, yes ...
Been using it for a call center since 2005.
Julian
2009/7/12 Alex Balashov abalas...@evaristesys.com
For 50 seats? I think so.
gergis.rasmy wrote:
i am asked to implement a call center of 50 seats for my company , and i
was wondering if Asterisk can fit this as a relaibale
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
Lyndon-Smith
Sent: 02 July 2009 17:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
Try client instead of component.
Make sure that you selected the component in the menu select as well
I can assure you that it works, and that it works well. We use it ;)
Julian
jonas kellens wrote:
I have installed gnutls and gnutls-devel from RedHat repositories
[r...@asterisk asterisk]#
usetls=no
Julian
jonas kellens wrote:
On Mon, 2009-07-06 at 16:18 +0100, Julian Lyndon-Smith wrote:
I can assure you that it works, and that it works well. We use it ;)
My jabber.conf :
[general]
debug=yes ;;Turn on debugging by default.
autoprune
I am using 1.4, and have the above device, and it worked really well
with monitoring 18 hints aka devices.
Now, I've moved us to a hotdesking paradigm where the user is the
extension not the device. IOW if I dial 1234, I will get user 1234
(who happens to log on to device ABC today, and DEF
I have a GXP2010 phone, the one with 18 blinky lights ;)
I currently provision the phone by writing out the conf file, encoding
it and sending it to the tftp server. I was wondering if anyone had
managed to automate the web side of provisioning ?
TIA
Julian
We have several types of phones,
cisco 7940/7960
aastra 55i/9113i/
grandstream gxp2010
I want to be able to implement hotdesking where an agent will logon to
any phone. I got all of that working, without having to reboot phones,
but then hit a brick wall.
Voicemail.
I still want each phone
Can I run two instances of asterisk sharing a single te412p ?
I want to be able to have several asterisk servers (for testing various
scenarios) running on one server. I was wondering if these asterisk
processes could share a zaptel/dahdi card nicely.
Julian
Eric Chamberlain wrote:
On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote:
Eric Chamberlain wrote:
[snip]
Thank you, that bug does have useful information.
We are working on moving from res_config_odbc to res_config_curl, so
all asterisk requests go through our django backend
Eric Chamberlain wrote:
Is there any documentation that explains res_config_curl?
We use the 1.4 backported version - it works so well I just can't sing
it's praises enough. We use it for realtime voicemail and realtime
queues / queue members.
Have a look at bug #11747 for some
See comments inline:
Steve Edwards wrote:
On Fri, 10 Apr 2009, James A. Shigley wrote:
Here is more or less what I'm trying to accomplish
1. Call comes in Plays Greeting
2. Starts Survey
3. Ask Q1, Record the answer (voice responses) repeat this step for
each Question
4.
Let me turn the question around slightly:
Are there any circumstances under which the h extension _won't_ get run ?
Julian
Julian Lyndon-Smith wrote:
Steve Edwards wrote:
On Sun, 29 Mar 2009, Julian Lyndon-Smith wrote:
Steve Edwards wrote:
Please show us the output from
I eventually found the problem - the h extension was getting run on the
Zap channel as soon as the bridge between the SIP client and Zap client
was broken. This is because of changes made to the cdr code in 1.4
trunk. However, the problem would not manifest itself to anyone except
those using
Asterisk 1.4 r181990
given the dialplan snippet below, can anyone tell me why the h exten is
not being run ?
console output:
[Mar 29 10:33:49] -- Executing [...@questionnaire-menu:1]
Set(Zap/1-1,
Meh. Has anyone got any clue ? I'm trying to test this tomorrow and it
is obviously not going to pass ;)
I've replaced the include = with a h,1,NoOp(here) and verified it with
a show dialplan
but that didn't work either
Julian
Julian Lyndon-Smith wrote:
Asterisk 1.4 r181990
given
Steve Edwards wrote:
Untopposting...
Ouch. Sorry.
Julian Lyndon-Smith wrote:
Asterisk 1.4 r181990
given the dialplan snippet below, can anyone tell me why the h exten
is not being run ?
This is not a dialplan snippet, this is the console output.
Yup, got it the wrong way around. Sorry
than duplicate code it is much better to
include it.
Julian
On Sunday 29 March 2009 11.42.29 Julian Lyndon-Smith wrote:
Asterisk 1.4 r181990
given the dialplan snippet below, can anyone tell me why the h exten is
not being run
Steve Edwards wrote:
On Sun, 29 Mar 2009, Julian Lyndon-Smith wrote:
Steve Edwards wrote:
Please show us the output from dialplan show questionnaire-menu.
Here you go
show dialplan questionnaire-menu
[ Context 'questionnaire-menu' created by 'pbx_config' ]
'0' =1. Goto(s
Can anyone give me any idea on where to start looking for this ? 1.4
svn (ish) It has appeared twice in the last hour on a system that gets
numerous inbound calls to the same number
TIA
Julian
[Mar 27 17:21:07] WARNING[3239]: ast_expr2.fl:407 ast_yyerror:
ast_yyerror(): syntax error:
Got this in the log, with no calls active. Is it a problem with my isdn
line, or * ?
[Mar 15 11:36:18] ERROR[29161]: chan_dahdi.c:8735 dahdi_pri_error: ACK
received for '0' outside of window of '39' to '40', restarting
[Mar 15 11:36:18] == Primary D-Channel on span 1 down
[Mar 15 11:36:18]
-
-- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith
-- Sent: 12 March 2009 10:21
-- To: Asterisk Users Mailing List - Non-Commercial Discussion
-- Subject: [asterisk-users] UK ISDN-30 and ANI
--
-- Has anyone in the UK got ANI to work on an inbound call ?
--
-- Using asterisk
David Quinton wrote:
On Thu, 12 Mar 2009 10:21:06 +, Julian Lyndon-Smith
aster...@dotr.com wrote:
Has anyone in the UK got ANI to work on an inbound call ?
Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30
AFAIK (and our E1 doesn't go to * box)
a) you mean
1 - 100 of 425 matches
Mail list logo
