Hi List
This may be a silly question by web searches etc don't seem to answer it.
Is there a CLI command to display ALL channel variables - standard and user
created - for a specific channel?
something like show channel SIP/Test123 all
I'm using Version 1.4.33.1
PG
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nal Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy
> > Grice
> > Sent: Wednesday, May 11, 2011 11:49 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] CLI -
Hi All
I am looking for a small scale Email to fax solution
Searches seem to throw up
AsterFax http://sourceforge.net/projects/asterfax/ which seems to go to
http://www.noojee.com.au/products/noojee-fax/fax-overview/
email12fax http://wpkg.org/email2fax/index.php/Main_Page
I would appreci
On 06/08/2011 01:09 AM, Paddy Grice wrote:
> Hi All
>
> I am looking for a small scale Email to fax solution
>
> Searches seem to throw up
>
> AsterFax http://sourceforge.net/projects/asterfax/ which seems to go
> to http://www.noojee.com.au/products/noojee-fax/fa
Hi there
I hope someone can help - I am having a big problem getting calls cleared
from several asterisk systems when RTP timeouts occur.
It appears that asterisk doesn't send a BYE when it decides to terminate a
call because of a RTP Timeout - is this a configuration problem? if so what
need cha
Hi
I use ADSL and SDSL on a lot of multi channel VoIP connections SIP and H323
and no real problems if you size the link correctly - this normally means
limiting no of calls to match available bandwidth.
Check out the upstream and downstream data rates and size on the smaller -
normally the ups
Hi all
A quick question about busy lamps
I have lamps working 'sort-of' on my GXP2000 lamps flash with ringing and
go solid red when call gets answered but stay green when a call is made from
the extension.
Setup is Ext 200, 201, 202, each monitor the other two
when 200 calls 202 - the BLF on
Rob Many thanks for the pointer - I was missing limitonpeers=yes in the
general section - Sorry I didn't say version (1.4.33.1) etc forgot with
frustration ;-)
Paddy
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Hi list
I have a requirement that I just don't know how to address - I don't think
its strange but can't find any pointers anywhere.
I have a user that wishes to have a "multi phone" divert. By that I mean
"calls made to his extension say Ext200 can be redirected to a different
extension say Ex
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy Grice
Sent: 19 August 2010 08:21
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Calling Line Identity - any ideas
Hi list
I have a requirement that I just don&#
long
as the trunk you're sending calls to the outside world over has the CallerID
setting set AND probably sendrpid=yes...in the sip configuration for both of
those items...past that, I could dig a bit
Cheers,
Sherwood McGowan
On Thu, Aug 19, 2010 at 2:25 AM, Paddy Grice wrot
Hi Sherwood
I actually do want "dynamic" CLID as I tried to make clearer
>> I don't know if this makes it any clearer -
>>
>> An internal call from Ext123 should send 123 as the CLID to SIP/Ext400
but should
>> send 442071110123 to SIP/TheWorld but an external call from
44123455667788 should
Hi All
Thanks for the pointers - I now have a working solution using local channels
and for the few occasions this needs to happen, about 300 calls in the
20,000 we handle each day I am very happy.
Again thanks for you help
Paddy
_
From: asterisk-users-boun...@lists.digium.com
[mai
Hi all
Been looking to find a way to stop the dtmf keys * 0 and # managing call
flow in the dialplan - I just want VM to stop recording on silence or
hangup.
I know I can trap the exit and loop back around but just want to ignore the
keys totally.
Any suggestions
P
--
oops - forgot to say this is voicemail() on Version 1.4.33.1
P
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy Grice
Sent: 02 September 2010 13:32
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voicemail
Hi List
I have had a look at the various posts on this and seem to be more confused
than ever - but then again that's not hard ;-)
I am using Version 1.4.33.1 build from the Debian "lenny" distros
I am trying to implement a simple screening
[macro-screen]
exten => s,1,Background(press1)
ex
Hi List
I am having trouble running the command
siptest:~# asterisk -rx 'dialplan reload'
most times it does what I expect and I get a response as below
siptest:~# asterisk -rx 'dialplan reload'
Dialplan reloaded.
every now and then I get no response i.e.
siptest:~# asterisk -rx 'dialpl
] Asterisk -rx command not returning data -
Version 1.4.33.1
On Mon, 14 Mar 2011, Paddy Grice wrote:
> I am having trouble running the command
>
> siptest:~# asterisk -rx 'dialplan reload'
>
> I assume the problem is timing but any ideas on how to fix it
I'm just a
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen
Sent: 28 May 2014 16:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 'restart when convenient'
asterisk-users-boun...@lists.digium.
Hi All
I am trying to get voicemail switched over to ARA on version 13 and notice
that the password is not stored in the db when it is changed.
A little hair pulling and playing around and I think the problem is in the
function ast_update2_realtime in main/config.c.
Issued source is ==>
int
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olli Heiskanen
Sent: 03 January 2015 08:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk removes a charachter from sip peer name
Hello a
roup.
This seems to be a real shortcoming in app_queue.
Any ideas, suggestions, anyone want to work with me to sort this ?
Paddy Grice
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name and the ruleset to apply from queuerules.conf?
On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice wrote:
Hi All
I have been looking at this problem for a few days/weeks now and after some
advice please.
I currently have a customer on 11.25.3 and I am in the process of upgrading
versions
usies);
}
So the penalties get calculated during the 'ringall' strategy and allowing
the queue app to exit, looping and raising the max penalty and calling the
queue app again.
Leon
On Thu, 29 Nov 2018 at 18:24, Paddy Grice wrote:
Hi John
This works f
Hi All
This sounds just like a problem I have had and still investigating having
moved to 16.9 using chan_sip. I am still trying to repeat the problem it
looks from debug that the issue is either voicemail of call transfer but I
cant consistently repeat it.
Voicemail is using ODBC and I just i
ery call some of which are simple sip
devices and others have to be local devices (Internal and External CLIs).
Paddy Grice
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Check out the new
&SIP/102&SIP/103&SIP/104&SIP/105
Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111
Exten => set1,1,Dial(SIP/112&SIP/113&SIP/114&SIP/1015&SIP/116&SIP/117
On Fri, May 1, 2020 at 3:22 AM Paddy Grice wrote:
Hi all
as
the same time so dial strings are very long
I
cant really use a queue(ringall) which was my original idea as the
customer
needs different groups for virtually every call some of which are simple
sip
devices and others have to be local devices (In
Hi All
I have a problem with queues that I have been trying to solve for many
months - the customer has now picked back up onto this and wanting a
solution - any guidance, ideas or solutions welcome.
This is the situation :-
We have a number of agents in a ringall group, a call joins th
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