ollider:out_1)])
> same => n,Hangup()
> [our_agi]
> Exten => 101,1,Congestion
> Exten => h,1,AGI(test.py)
>
>
>
>
>
>
> On Thu, Oct 15, 2020 at 12:35 PM Peter van Haaften
> wrote:
>
>> Hi,
>>
>> I am trying to write a dialplan that will us
Hi,
I am trying to write a dialplan that will use Dial() to call two local
extensions. One extension will run an AGI script (a continuous background
process, running until hangup), the other will connect the active channel
to Jack() (also running as continuous process, until hangup).
This is my
On Montag, 27. August 2018 17:42:37 Hans-Peter Jansen wrote:
>
> What am I missing here, any suggestions?
>
Okay, scratch it, "notifycid = yes" must reside in the general section!
Now, it behaves as expected until:
[Aug 27 22:20:37] NOTICE[6200][C-0003]: app_di
Hi,
while trying to get my new Asterisk 15.5.0 PBX replacing a 11 years old
Asterisk 1.2.31 ISDN BPX, I'm stuck to get call pickup going as usual.
The old one uses specific patches, IIRC...
If I interpret various sources of related information correctly, current
Asterisk versions should
Hi,
is somebody attending, that wants to share his outgoing dial rules of
extension.conf, like used in typical(?) german pbx setups?
* zero prefix for outside calls
* zero zero or plus prefix for international calls
* handle emergency calls
With ISDN, one was able to just forward the called
Hi Daniel,
On Montag, 5. Juni 2017 21:45:01 Daniel Tryba wrote:
> On Mon, Jun 05, 2017 at 06:10:50PM +0200, Hans-Peter Jansen wrote:
> > ; matches 12345678099, too
> > exten => _1234567800,1,Dial(SIP/int)
> >
> > Except from SIP invite with tcpdump:
> >
>
Hi,
I just started with setting up a new asterisk system, that will operate on a
sip trunk, but I wonder, how to transfer the calls to different extensions,
because all calls appear as being send to the base number of the trunk.
E.g. given the trunk range of 1234567800-12345678099, a call to
ruary 2016 at 10:29, Peter Wallis <pwal...@acm.org> wrote:
> I am having real trouble getting started. A definitive "hello world" is
> certainly missing from the official site and the ones out there are dated
> or broken.
>
> I am beginning to think something went
I am having real trouble getting started. A definitive "hello world" is
certainly missing from the official site and the ones out there are dated
or broken.
I am beginning to think something went wrong with the install. It was a
fresh install of an Ubuntu server, and a fresh install of 13.7.0
If anyone else is having this issue, asterisk 1.8.32.3 uses the Path header
as expected, if you want to follow progress I've created a bug report:
https://issues.asterisk.org/jira/browse/ASTERISK-25666
On 6 January 2016 at 10:29, Peter Baines <li...@pbaines.com> wrote:
> Hi,
>
&g
PBX 13.6.0.
c=IN IP4 192.168.68.68.
t=0 0.
m=audio 12356 RTP/AVP 0 8 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=maxptime:150.
a=sendrecv.
Thanks,
Peter
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Hi Guys,
Just new to Asterisk and am completely stumped. I have created two accounts
as instructed. Please see below for the config of the user accounts.
[Peter]
type=friend
host=IP address
disallow=all
allow=ulaw
allow=alaw
callerid=Peter 6004
secret=XXX
context=default
, April 16, 2014 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FW: clients unable to auth
asterisk-users-boun...@lists.digium.com wrote on 04/16/2014 05:56:32 AM:
From: Peter Reid peter.r...@morodo.co.uk
To: asterisk-users@lists.digium.com
Hi,
I searched a lot last few days but I am uanble to find a DID number in
Macedoania.
However no luck. any ideas about a provider ?
Thanks,
Peter
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On 04/09/2013 19:31, Markus wrote:
few years while this country progresses. But you can always get a
premium number there (pay per minute/call) if that helps.
Thanks :-) Do you know any good premium provider there ?
Peter
I had this exact problem with my voip provider a few years ago.
It was disconnecting at exactly 5 minutes.
I solved it by moving Asterisk 1.6 to Asterisk 1.4.
Try asterisk 1.4 or 1.8 on a test box and see how it goes.
Peter
On 21/03/2013 09:31, Florian Wolters wrote:
Hi @ll,
I just moved
from ssh(console) run which php
it should give you path where it is installed.
Peter
On 30/01/2013 08:28, Muhammad wrote:
Hi,
I used elastix with asterisk 1.8
when I run my AGI code, cli give me theses errors:
SIP/147-0098AGI Tx agi_callingpres: 0
SIP/147-0098AGI Tx
it inside asterisk ?
Thanks,
Peter
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--
Hyeroba Wegulo Peter
Managing Director
Jezitech Solutions Limited
P.O.Box 36285,
Plot 27,
Sir. Apollo Kagwa Road
Kampala Uganda.
website: www.jezitech.com http://www.jezitech.tk
Phones : +256-414-533238
I have been looking online for a definitive how-to on using Asterisk as a
SIP trunk termination point . I am seeing conflicting messages and
methodologies for doing this.
I am not going to use a commercial vendor for this trunk, it will be used in
testing out various customer scenarios and am
in advance.
Best regards,
Peter Gelencser
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Hello all,
I am new to the Asterisk IRC users group. I was wondering if it would be
possible to use an IRC client when reading through the posts. If so, can
someone recommend one, and how I should go about configuring the client.
Thank you for your time and assistance.
Peter
where I cant find the ipaddr field. ) , the CLI shows me an error
message: Unable to get IP address of peer 'extension_number'
So its quite confusing. Why it dont fill the ipaddr field?? From which SIP
message get and cut out the IP address?
Thank you
Peter
[57@default:1] Record(DAHDI/4-1, 0620XXX-57.wav) in
new stack
-- DAHDI/4-1 Playing 'beep.alaw' (language 'en')
What do I wrong? Should I set any other parameter than this? Thanks for
you help in advance.
Best regards,
Peter Gelencser
start. Dropping.
Thanks for your help in advance.
FYI am using Asterisks 1.6
Peter.
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signalling to ss7 in dahdi_channels.conf
How do I sort this out?
Thanks for your help in advance.
Peter.
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dials)
Exten = _X./103,n,hangup
Exten = _X.,n,Dial(DAHDI/1,w,5551212)
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Peter den Hartog
*Sent:* Thursday, March 24, 2011 3:45 PM
*To:* Asterisk
-Original Message-
From: Peter den Hartog peterdenhar...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 25 Mar 2011 09:14:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non
= _X.,n,hangup
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Peter den Hartog
*Sent:* Friday, March 25, 2011 3:15 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject
Hi,
I would like to use the from caller id, to allow calls yes or no.
101, and 111 should be allowed to use the Trunk, the rest of the phones are
not.
Is this even possible?
So if the from caller id is 101 or 111, then allow the call, otherwise
hangup.
Thanks,
Peter
PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Thursday 24 Mar 2011, Peter den Hartog wrote:
I would like to use the from caller id, to allow calls yes or no.
101, and 111 should be allowed to use the Trunk, the rest of the phones
are
not.
Is this even possible?
So if the from
Hmm disabled Woomera and everything seems stable.
Strange!
On Thu, Mar 10, 2011 at 11:46 AM, Peter den Hartog peterdenhar...@gmail.com
wrote:
1.8.0 :-), Nothing fancy just simple dialing/trunking.
On Thu, Mar 10, 2011 at 11:31 AM, --[ UxBoD ]-- ux...@splatnix.netwrote
me that, right?
Thanks,
Peter
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asterisk
? I tried to debug it but i couldn't find
a good reason for the crashes.
Maby the box is just overloaded or something like that but there should be
a log file telling me that, right?
Thanks,
Peter
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Rsync to sync /etc/asterisk and use keepalived/heartbeat for a failover
Asterisk IP.
Make sure to read this -
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+SolutionsFor
From IP rewrite
On Mon, Feb 28, 2011
I'm running Asterisk 1.6 and was wondering if anybody have a workig barge
in solution running.
I was thinking of using chanspy, but i would like that the original call
would be dropped, and the new call would be the only one there.
--
Okay, so let me try to make it more clear to be sure everybody gets it :-),
i can be a bit unclear from time to time ;-).
100 is in a call with 101.
102 has a higher priority and calls 100. The call between 100 101
disconnects, and 102 100 are connected.
Peter
On Wed, Feb 16, 2011 at 11:29
on this issue? Would be great!
Thanks,
Peter
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Anyone has the same problem or is it just me?
Please give me some hint.
Thanks,
Peter
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callerNametrue/callerName
callerNumberfalse/callerNumber
redirectedNumberfalse/redirectedNumber
dialedNumbertrue/dialedNumber
/forwardCallInfoDisplay
/line
Thanks,
Peter
From: Cassius Smith [mailto:cass...@cassius.org]
Sent: Monday, November 22, 2010 1:12 PM
To: kowalla...@gmail.com
Solved!
Thank you Jonathan.
Like you suggested - I've changed port on both lines to 5060 and changed
contact so all: name, authName and contact are the same and it is working
like charm.
Thanks again,
Peter
-Original Message-
From: jthurma...@gmail.com [mailto:jthurma...@gmail.com
problem like this?
Any hint would be appreciated.
Peter
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http
? Are we talking up to around 5
agents or something? Is there a limit on the number of queues?
(I'm sure there is a page on the website that answers most of these
questions, heh :))
Leif Madsen.
See http://queuemetrics.com
Peter
We're mostly Cisco CallManager with some SIP and Asterisk.
I want someone at one of our locations to be able to dial and number
and have Asterisk simultaneously dial several Call-Manager extensions
which are set to auto-answer and talk into the phone creating a sort
of paging system.
We have
, but I've not
heard any method to do this.
Peter.
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in to the phone and listen
in to everything, ie bit like call monitoring but include the bits
between the calls
Does anyone have any ideas, or is it going to be quickest to write my own.
Peter.
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the returned list using
cut searching for the device.
So.
1. Is there a function I'm missing to do this say..
is_queue_member(queuename,channel)
or
2. Is there some way of creating such a function.
Thanks in advanced
Peter Childs
On 11 July 2010 14:19, Paul Belanger paul.belan...@polybeacon.com wrote:
On Sun, Jul 11, 2010 at 5:40 AM, Peter Childs pchi...@bcs.org wrote:
1. Is there a function I'm missing to do this say..
is_queue_member(queuename,channel)
*CLI core show function QUEUE_MEMBER
No function by that name
On 11 July 2010 15:56, Paul Belanger paul.belan...@polybeacon.com wrote:
On Sun, Jul 11, 2010 at 9:28 AM, Peter Childs pchi...@bcs.org wrote:
No function by that name registered.
also its not listed on voip-info, I'm using SARK/Asterisk 1.4.21
The function is in 1.6.2. Best you could do
options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Groet // Kind regards,
Peter den Hartog
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make[1]: Leaving directory `/usr/src/linux-2.6.32.13'
make: *** [all] Error 2
Do you have any clue what is the problem and how to solve it? Thank you
for your help in advance.
Best regards,
Peter Gelencser
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=strict, leavewhenempty=strict
Using Asterisk 1.4 and a Sark 850.
Any help, or at least where to go
Peter.
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tie the two concepts together.
Peter
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Unfortunately no, it did not solve my problem, the sitation is the same.
Any other hint?
Best regards,
Peter Gelencser
2010.05.01. 9:38 keltezéssel, Rudi Oosthuizen írta:
Had a similar problem with a B410p BRI card. Had to enable (or disable)
the 100ohms termination jumper on the card
.
Thanks in advance.
Peter
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asterisk
Hi,
I have one in stock - got it from a client who wanted to get rid of all
his old IT equipment.
Looks strange, did not have enough time to play with it Tried it
once, looked hard to configure.
It stays unused in the storage room.
Peter
On 29.4.2010 10:20, Tim Nelson wrote:
Greetings
and wait 40 sec, the card can connect and works properly.
The telco says the asterisk crashes the connection with the telco, when
I let the NT reconnect, the card connects properly.
Do you have any idea how to solve this problem? Thanks for any help in
advance.
Best regards,
Peter Gelencser
for you help in advance.
Best regards,
Peter Gelencser
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2010.04.20. 16:50 keltezéssel, Shaun Ruffell írta:
On 04/19/2010 03:48 AM, Peter Gelencser wrote:
I've run into a veird problem. I'm using a B400P BRI and an A1200P card
with dahdi (2.2.1) driver. The dahdi_scan shows the each moduls and
spans, everything seems fine. With dahdi_genconf I
motherboard, the situation is the same so it's not
chipset specific.
Please let me know if there is any solution. Thank you for your help in
advance.
Best regards,
Peter
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to 1.4 and issue
got solved.
Never managed to find what is going on.
It was happening only if all were true:
- linksys phone or pap
- asterisk 1.6
- use certain VOIP provider.
Solution: moved to 1.4
I hope thsi helps.
Peter
.
Its not about money its about evangelism.
Peter.
On 03/29/2010 07:48 PM, Alex Balashov wrote:
Sounds like the church has strayed from its core competencies and
invited the money-changers into the temple.
Being the official asterisk-biz harbinger of God's wrath, I suggest an
intensely
in his
e-mailbox on his direct number.
Any pointers would be highly appreciated!
Thanks,
Peter
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, 2010 at 5:40 AM, Peter den Hartog
peterdenhar...@gmail.com wrote:
Hello,
I was wondering if the following was possible:
When somebody sends a fax to my direct number 0101234567105 (my extension
will be 105) is it possible that Asterisk, or an addon sees this as a
fax,
and e-mail the fax
On 11 March 2010 21:09, Matt Riddell li...@venturevoip.com wrote:
On 9/03/10 9:13 PM, Peter Childs wrote:
Also is there some way to get the starting end to auto pickup, (or at
least hit for this to happen (I'm using SIP if that helps))
When you make an originate request it works like
On 8 March 2010 15:34, Olle E. Johansson o...@edvina.net wrote:
8 mar 2010 kl. 11.13 skrev Peter Childs:
On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote:
At an Asterisk CLI use the command manager show commands.
Life is rarely that simple, and this does not really answer
On 9 March 2010 07:58, Peter Childs pchi...@bcs.org wrote:
On 8 March 2010 15:34, Olle E. Johansson o...@edvina.net wrote:
8 mar 2010 kl. 11.13 skrev Peter Childs:
On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote:
At an Asterisk CLI use the command manager show commands.
Life
command means a Call to play the DTMF on,
where as Channel in a Originate command means the Device to place the
call on so you can't use the same input for both commands (or can
you?)
Peter
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standard.
Peter.
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asterisk-users mailing
Hi,
How would you like it to work? It would be an inner extension or this
fax should be reached from a public phone number?
Best regards,
Peter Gelencser
2010.03.05. 13:06 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta:
Hi Experts,
I have an asterisk machine with DAHDI and i want
As far as I know, you should set up the callerid in the chan_dahdi.conf
with the usecallerid=yes and the callerid=8001234001 options where you
are setting the each channels.
Regards,
Peter Gelencser
2010.03.05. 7:54 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta:
Hi All,
Finally I
Then it's simple, set up an exten in the extensions.conf like
exten = 123456789,1,Dial(DAHDI/2,,rtT)
exten = 123456789,n,Hangup()
replace the 123456789 with the public phone number and the DAHDI/2 with
the channel you are using.
Regards,
Peter
2010.03.05. 14:07 keltezéssel
all trunk groups and users, only using G711
at this point.
Peter
On Sat, Feb 27, 2010 at 10:05 PM, Tri Tu mtr...@yahoo.com wrote:
RTP is only firewall issue. Make sure that you can pass traffic from your
client to the asterisk server. If it's on the same LAN, there shouldn't be
any issue
issues, like trying to negotiate G729 when it's not
capable, but since then, I've changed everything back to G711.
I have connected to it, a SIP trunk, 3 registered users and I'm at a loss as
to how to troubleshoot this further.
Can anyone point me in the right direction?
Peter
--
Peter Serwe
http
context=default
Please let me know what do I misconfigure. Thanks for any help in advence.
Best regards,
Peter Gelencser
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miss? Thanks for you help in advance.
Best regards,
Peter Gelencser
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that's why there is an 1...@100)
I hope anybody has any input on this because i'm lost :-) never had this..
it's just a simple dial..
Thanks,
Peter
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of opensips and it worked..
thanks for the input tho :)!
Peter
On Wed, Feb 10, 2010 at 2:44 PM, Kevin P. Fleming kpflem...@digium.comwrote:
Peter den Hartog wrote:
Hello,
I installed Asterisk in a linonde cloud debian 5, and i'm trying to
create a first call but when i try to set up the call i
for the price
Peter
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SPA504G is LINKSYS with newer look and HD :-)
Expect all you had in Linksys SPA9XX + more.
I personaly have both phones - differences are not a lot :)
Peter
On 10.2.2010 20:31, Jeffrey Ollie wrote:
On Wed, Feb 10, 2010 at 12:23 PM, Brent Torrenga li...@torrenga.com wrote:
Coming from
and one that is made by Eurodesign.
The one from portech is like a trunk while the one from eurodesign
relies on USB and project GSMOPEN.
what would you recommend - trunk or usb ? Or there are other
possibilities ?
Thanks,
Peter
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
We use this one, and it works great.. easy to setup and it works with a
normal network connection :)
On Mon, Feb 8, 2010 at 1:52 PM, Peter peterp...@aboutsupport.com wrote:
Hello,
I am looking for a gsm
Hi,
I can not find pricing and shipping information for these. I tried to
contact their sales for these. We will see, but most likely we will go
with portech.
Peter
On 08.2.2010 15:15, Peter den Hartog wrote:
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway
, Queue's etc etc.
Any pointers on how to get started would be most helpful.
Peter.
---
(sorry this is so long)
Peter,
I figured that I would chime in, as I run IT and am a managing partner of a
small call center based
...@gmail.comwrote:
On Mon, Feb 1, 2010 at 8:55 AM, Peter den Hartog
peterdenhar...@gmail.com wrote:
I'm using the default Asterisk function Monitor to record calls, but i
have
some issue's with this, the problem is when a call is finished, it never
mix
in out together, bellow you can see
ex...@exten.
Even on a normal Asterisk machine, i have issue's with recording, i'm using
Asterisk 1.6.2.
Anybody got any tips on this?
Thanks,
Peter
--
Groet // Kind regards,
Peter den Hartog
Sent from Amsterdam, NH, Netherlands
.
Peter.
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quickly.
Peter
option for your endpoints?
y.
Peter Childs schrieb:
Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash
I've managed to get a basic system set up. and can now take and make
sip calls over the sip trunk I've got from sipgate.co.uk for testing
purposes
2010/1/26 Peter Childs pchi...@bcs.org:
2010/1/26 Yves Arikoglu yves...@gmx.de:
do you use the
qualify=yes
No, If I do it does not work at all.
I've found if I set defaultexpiry to 30 it works fine. and was infact
working for 30 seconds every two minutes before, It looks like
Somewhere \n needs to be converted into utf8 new line. Asterisk should
do this for you but it doesnt.
Try opening the dialplan in hex mode and insert hex code for utf8 new
line where the line break should be.
Peter
On 17 jan 2010, at 12.09, Dominik wrote:
Hello,
I'm using Asterisk
he who has just installed Ubuntu over the network to check the
computer works!)
Peter.
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be most helpful.
Peter.
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problem but is
trying to see if he can get it to work. That's how bad the Alcatel
phone system is!
Peter.
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.) but it has its benefits, such as not being restricted by a
particular GUI or management system, and being able to customise things
a bit more.
Peter.
Rob
On Tue, 2010-01-12 at 10:55 +, Peter Childs wrote:
This is currently still at a proof of concept stage.
After being mis-sold
to me in various situations and is usually a result of
reverse DNS lookups failing on the MySQL side. See:
http://dev.mysql.com/doc/refman/5.0/en/dns.html
for a solution to that if that is the issue.
Best regards,
Peter Lindqvist
Voxion Ltd.
www.voxion.net
You may want to make your WaitExten() wait for a bit longer than 3
seconds. ;) Doesn't give the user much time to listen to the greeting
AND enter a 4-digit extension
I have something like this in my dialplan and it works fine (sanitized,
and some irrelevant parts snipped out). I
Try removing the include statements from the default context and see
what happens. Also double check to make sure calls are sent to the
default context.
Peter
On Nov 2, 2009, at 3:40 AM, Thomas Person wrote:
I want to match specific contexts to menus.
If users dial a number (example
Check the channel driver configuration file, or fire up CLI with max
verbosity and monitor its output while calling the dialplan
extensions. CLI is like a good friend that tells you whats going on
and if there are any errors in you configuration.
Peter
On Nov 2, 2009, at 4:39 AM, Thomas
Hello. I think i've seen this problem, it was generated by a missing ACK
on 200 OK. If that is the case try modifying session timer parameters in
sip.conf so a missing ACK will not lead to call termination.
Peter
Ishfaq Malik wrote:
Hi
I'm having an issue with just one of the phones
the call?
peter
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