Displaying 20 results from an estimated 10000 matches similar to: "UK ISDN-30 and ANI"
2009 Jan 13
2
0800 UK number
I have concocted a system for my children's primary school where parents
can dial in and subscribe to an "emergency broadcast" message so that
they can be automatically contacted in case of a problem like the school
being shut because of snow etc.
I would like to provide an 0800 number service for this, so that there
is no cost to the parents, but obviously I would like to get
2008 Dec 16
4
RDNIS and asterisk
I have a couple of numbers that are diverted to a number that is
conected to an isdn30 card, running asterisk 1.4.
eg.
123456 => 22334455
654321 => 22334455
What I would like to know is the number of the orginal number dialled
(123456 or 654321). I thought that RDNIS was the answer, but it is
always coming up blank.
When I did a debug on the pri span, I saw the following message
2009 Feb 12
4
Multiple caller id ...
If I have the following in the dialplan
exten => foo,n,Dial(SIP/1234&Zap/G1c/55443322)
and SIP/5432 calls this extension,
is it possible to show different callerid numbers to each of the target
numbers ?
The reason I ask is that if the call is from an internal sip phone, I
want to show the internal callerid (5432) to the SIP phone on 1234, and
the DDI of the 5432 extension
2007 Mar 19
3
Cepstral and numbers
Does anyone have any idea on how to force cepstral to convert a number
to speech ?
I have noticed that sometimes it speaks the number correctly, and at
others it doesn't.
1) 787 is pronounced 7-8-7
2) 123 is pronounced one-hundred and twenty-three.
1) is wrong for what i need, 2) is perfect.
Is there anyway of forcing numbers to be pronounced as 2) ?
I've tried looking at the ssml
2009 Jul 02
3
Grandstream 2010 and blinky lights
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 tomorrow).
Can I make the GXP monitor user 1234, not extension 1234 ?
2008 Oct 01
3
GSM / 3g channel bank
More than 60% of our outbound calls are now to mobiles, so the time has
come to whack in a gsm channel bank.
Does anyone have any preference of bank ? Do you use a PRI or VOIP
connection from the bank to asterisk ? Real-world experiences are sooooo
much better than marketing blurb ;)
We currently have a TE412P with a free socket, so we have a choice
either way. I am looking for up to 30
2007 Jul 28
3
global variables and updates
Sorry if this appears twice - I originally sent it nearly 18 hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to make sure I don't put two people
together !)
I was going to use a global variable ${NEXTMEETME}, and add one every
time I
2007 Aug 24
1
Simulating errors (Busy / Out of Order)
I'm trying to build a test suite so that I can run "calls" through and
verify the call results.
I've made a cross over cable and linked my 2 ISDN30 ports together. So
now I can dial out on span 1 , and to receive the call on span 2.
in the context for span 2, I have the following:
<snip>
; #1 "answer" a call and play music
000XXX : ring for a random period,
2007 Oct 11
9
Mask Initial Processing with Ring Back Tone
I need to process a number of lines of code in the dialplan before answering a
call. Can standard ring back tones be played to the caller while this is
happening prior to answering the call. Which commands would facilitate this?
Thanks in Advance,
Vic
2006 Mar 15
3
Double-ring tone
I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works
fine. Except that when I make an outbound call, I get a double-ring
sound. I also found that if the target number is engaged, I get a ring
sound and at the same time get a busy sound.
If I revert back to 7-4, there is no problem.
Anyone else had this, or any clues on how to fix it ? All of our other
phones are still on
2007 May 24
3
meetme sounds
I am playing around with dynamic meetme conferences, and wanted to have
one person constantly in the conference, with calls "popping in and out".
Is there an option / any way of playing enter / leave sounds to the
person who created the conference only, and not the people leaving /
joining ?
TIA
Julian.
2010 Aug 28
1
Play a number of files to a caller
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), and when
finished, go back to the music.
1) I thought of redirecting to an extension that played the
2012 Nov 07
1
Random crash of the machine ? due to Asterisk 11
I experience random crash of machine (full hang, requiring a hard reset)
after trying to test run Asterisk 11.
The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled
from the source and no other software has been installed
Anyone experience similar situation?
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2006 Jun 13
3
Queues and macros and agents
When a caller in the queue is connected to an agent, the call is placed
to the extension and context specified using Agentcallbacklogin. This
allows for me to add extra things to the diaplan *before* calling the agent.
Now, I want to be able to use a device, rather than agents. So I can use
addQueueMember and add my SIP device. However, I still want to do a
couple of things before the device
2006 Jun 14
2
AddQueueMember and Local channels
Following on from a posting yesterday from Kevin, I have the following
in the dialplan:
exten => 709,1,AddQueueMember(SomeQueue|Local/706@AgentQ)
I am on extension 706.
From the CLI:
SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s
holdtime), W:0, C:0, A:3, SL:0.0% within 60s
No Members
No Callers
I call 709, get a console message
2008 Nov 12
3
Grandstream and pickup
Man, I really feel stupid, but after banging my head on a brick wall for
several hours ... I need help!
I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten
5707, and I've got an xlite on 5608.
When I make a call from an outside line, I dial SIP/5608. The little
blinky light on the GXP that's monitoring 5608 goes, well, "blink
blink". :) I then press
2009 Jan 16
4
Snom 300 vs Grandstream gxp
Can anyone who has used both comment on the pros and cons ? Need to buy
about 30 of these, for a small company with limited IT support.
Julian
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2006 May 31
2
AEL2 and CID
Does anyone know how to get CID working in AEL2 ?
In extensions.conf you can do:
exten => 111/666,1,PlayBack(demo-congrats)
exten => 111/666,2,Hangup()
exten => 111,1,PlayBack(demo-moreinfo)
exten => 111,2,Hangup()
and if callerid 666 dialed 111, they would get demo-congrats, everyone
else gets demo-moreinfo.
In AEL:
111 => {
Playback(demo-moreinfo);
2009 Feb 06
2
asterisk and DNS
We've just had the problem where our DNS server went down, and * started
to act "funny".
Is the best solution to install a local DNS server on the * box, and
have no other DNS servers ? - this is an internal app, no need for any
external DNS resolution at all.
Julian.
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