I would like to build into our application a button to park and a
button to unpark calls.
Consider this scenario:
Agent A gets a call, and obtains a reference number. He needs to send
this client to another Agent.
Agent A pushes the park button. The call is then parked into a
designated slot
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
Thanks for all the replies and help - For future reference I eventually
decided to go for a func_odbc solution and use a database sequence that
*is* atomic and gives me what I needed.
Julian.
Julian Lyndon-Smith wrote:
Sorry if this appears twice - I originally sent it nearly 18 hours ago
I'm trying to write a zap monitor program to visually display the status
of each channel. It's working well -:
However, one thing that I am still struggling with is knowing the status
of the zap channels when the program starts.
Zap show channels only seems to show an extension on an inbound
I'm trying to write a dialplan that will allow me to stress test it. I
want to be able to dial an extension, or pretend that the extension is
busy or out of order (so that I can see what to do)
given the dialplan snippet:
[outbound]
exten = _X.,1,NoOp(${TEST})
exten = _X.,n,Dial(SIP/${EXTEN})
Oh, for god's sake.
how stupid is I am feeling :)
My brain cell is feeling very ashamed.
Julian.
James FitzGibbon wrote:
On 8/3/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
why if I call the Busy or Congestion extensions, the DIALSTATUS and
HANGUPCAUSE variables are not set ?
If I
And what of all the folk that have a v1 card (I've got 2 quad-ports
sitting here) ?
And can you cross-ship a v1 card for a v2 card replacement ?
Julian.
Steve Totaro wrote:
Kevin P. Fleming wrote:
Stephen Bosch wrote:
The only way this will ever happen is if Digium completely
Kevin P. Fleming wrote:
David Gomillion wrote:
Last I checked, the replacement with the new firmware is only for those
who bought the card in the last year (i.e. the card is still under
warranty). Those of us who were early adopters cannot enjoy the
improvements of the upgraded firmware
I have a need to have a GUID (for example,
bcd47ccc-d7c9-ddb6-dc11-6746a770d77d [36 characters long including the
-]) generated in the dialplan. Is there any asterisk function that
would do this ? I would prefer not to have to shell out every time a
call comes in.
Julian
James FitzGibbon wrote:
On 8/9/07, *Julian Lyndon-Smith* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I have a need to have a GUID (for example,
bcd47ccc-d7c9-ddb6-dc11-6746a770d77d [36 characters long including the
-]) generated in the dialplan. Is there any asterisk
I am originating calls through the Manager Originate API command.
I can track failures (through the OriginateResponse event)
I can track answered calls through the OriginateResponse event)
There may be occasions where I need to cancel some outbound calls whilst
they are ringing.
Here's my
Rory Campbell-Lange wrote:
We are trying to configure a Sangoma A101 card to allow both incoming
and outgoing calls on a UK (BT) ISDN30e line with only 24 channels
enabled.
At present incoming calls work fine. We can't call out -- we get a
BUSY/CONGESTED error.
What is your dialling plan
I think that revision 80086 in the 1.4 subversion branch would fix this.
Julian.
Peder @ NetworkOblivion wrote:
Does anybody have realtime queue members working? Not the queues
themselves, just the members. I have realtime working for voicemail and
sippeers, but I can't get queue members
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
Oh, man, why is it that when you spend hours working on something, as
soon as you send a message for help, the solution presents itself ?
To answer my own question, and for prosperity, see the comments inline:
Sorry for the waste of bandwidth :(
Julian Lyndon-Smith wrote:
I'm trying to build
We have been running 1.4 since July 06 (it was trunk then), and have
upgraded often with the 1.4 branch (Currently on SVN-branch-1.4-r77571).
We have 100+ extensions (SIP) and 30 ISDN channels. We often have 50+
agents available for outbound calls and queues (20+ queues). We are
making /
I'm pretty sure that a command to reset the counters was added soon
after this patch.
Julian.
James FitzGibbon wrote:
On 8/24/07, Marlon Dutra [EMAIL PROTECTED] wrote:
Every queue has some status counters (completed, abandoned, hold
time...) that are very useful for statistics. The problem
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
We have been running 1.4 since July 06 (it was trunk then), and have
upgraded often with the 1.4 branch (Currently on SVN-branch-1.4-r77571).
We have 100+ extensions (SIP) and 30 ISDN channels
We've just received a bill from bt where it claims that we are making
numerous calls to the same number time after time.
e.g.
01226xx Barnsley20/06/2007 211516:00:00
01226xx Barnsley20/06/2007 121908:55:32
01226xx Barnsley
try
Nicholas Blasgen wrote:
I've got a macro that tries to find the first available SIP trunk to send
outgoing calls on. It tracks the usage of the lines (since each trunk has a
call-limit of 2) by using GROUP(). My problem is that once a call switched
to ANSWER state, ``group show
I have sent a few emails over the past couple of days that simply have
not arrived on the list (or so it seems).
Is anyone else encountering this ?
Julian
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Thomas Kenyon wrote:
Is there an advantage to having a Queue members URI in the form:
SIP/User (or indeed IAX2/User)
Over
Local/number@context
?
I know that the latter will allow you to do things like set counting
logic etc. through dialplan operations, but the former appears to be a
Nothing from me is posting to the list either.
Julian
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To UNSUBSCRIBE or update options visit:
Julian Lyndon-Smith wrote:
Nothing from me is posting to the list either.
heh. Thought that this trick would work: it did for Doug.
I've been trying to send the email below for 3 days now !
I know this is probably going to ignite the flames again ..
I have looked at the recent threads
Julian Lyndon-Smith wrote:
Julian Lyndon-Smith wrote:
Nothing from me is posting to the list either.
heh. Thought that this trick would work: it did for Doug.
I've been trying to send the email below for 3 days now !
I know this is probably going to ignite the flames again ..
I have
Thanks Matt,
Matt Florell wrote:
On 10/6/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Julian Lyndon-Smith wrote:
Julian Lyndon-Smith wrote:
Nothing from me is posting to the list either.
heh. Thought that this trick would work: it did for Doug.
I've been trying to send the email below
I'm trying to find the right Digium card for the
Dell 2950
Dell 2850
HP DL380 G3
HP DL360 G3
Are these 3.3v or 5.0v machines ? I am out of the office, and need to
buy a card today.
I am looking at either the TE407 or TE412, and would appreciate any help. :)
Julian
Just as a follow up on this thread, I decided to go for the Digium 412P
quad port card.
Thanks to everyone who commented, positively and negatively - it helped
provide a balanced view in the end.
Julian.
Matt Florell wrote:
On 10/6/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Julian
Russell Bryant wrote:
I have been having discussions with various members of the development
community
in regards to changes to the way we manage open source Asterisk releases. The
changes that we eventually decide on will determine how we manage the 1.6
version of Asterisk. I will be
I am *really* sorry about hijacking this thread, but the only way I can
post to the -user list is by replying to another thread. (btw, this is
getting really annoying. Please, Digium, sort the filters out!)
I installed my super-duper new TE412P card today, without remembering to
check the
I am *really* sorry about hijacking this thread, but the only way I can
post to the -user list is by replying to another thread. (btw, this is
getting really annoying. Please, Digium, sort the filters out!)
I've added the auto-answer header in my dialplan, and it works great.
However, there is
Hijacking a thread again - the only way I can post to the -user list is
by replying to another thread. (btw, this is getting really annoying.
Please, please, please, Digium, sort the filters out!)
I've added the auto-answer header in my dialplan, and it works great.
However, there is a problem
Eric ManxPower Wieling wrote:
Julian Lyndon-Smith wrote:
Hijacking a thread again - the only way I can post to the -user list is
by replying to another thread. (btw, this is getting really annoying.
Please, please, please, Digium, sort the filters out!)
You seem to be subscribed
Lyndon-Smith wrote:
I am *really* sorry about hijacking this thread, but the only way I can
post to the -user list is by replying to another thread. (btw, this is
getting really annoying. Please, Digium, sort the filters out!)
I've added the auto-answer header in my dialplan, and it works great
Apart from religious grounds (!), is there any pros or cons why I should
choose one over the other for a new install of asterisk ?
Julian
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asterisk-users mailing list
To
These modules are not part of the standard 1.2.3 release - did you also
install the 1.2.3 release of the asterisk-addons package ?
If * is loading older modules (which it probably is because of your
config files) then it may cause grief ;)
My .2p worth. Probably not helpful, but maybe, just
that
these modules are from a previous asterisk version.
Let us know how you get on.
Julian.
Dan Littlejohn wrote:
On 1/27/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
These modules are not part of the standard 1.2.3 release - did you also
install the 1.2.3 release of the asterisk-addons package
app_saycountpl.so
app_striplsd.so
app_substring.so
app_txfax.so
cdr_addon_mysql.so
chan_modem_aopen.so
chan_modem_bestdata.so
chan_modem_i4l.so
chan_modem.so
format_mp3.so
res_config_mysql.so
Warren Burstein wrote:
Julian Lyndon-Smith wrote:
These modules are not part of the standard 1.2.3 release
Does any one know how to change astmanproxy to be able to
a) compile without optimisations
b) dump a core
I've had it crash several times over the past couple of months, but
there is no way to debug what's going on.
I like the way a core is produced when (if!) * crashes, and would like
to
What does show config mappings show ?
Julian
Stuart Ford wrote:
Dear all
I'm writing to the list for help as a last resort. I've exhausted all
other options, so please forgive me. I've lurked here for years but
never actually posted.
I'm trying to get Asterisk Realtime SIP
Check the number of calls waiting in the queue, then play the message if
more than 0
example code (written in the TBird IDE)
Exten = 100,1,Answer()
Exten = 100,n,Set(NumWaiting=${QUEUE_WAITING_COUNT(${QUEUENAME})})
Exten = 100,n,GotoIf($[${NumWaiting} = 0]?JoinQueue)
Exten =
You could always ask IE to emulate firefox
;)
Sorry, couldn't resist ...
Julian
Tony Mountifield wrote:
When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays
a message at the top Your browser is not supported by this version of GUI!,
and We recommend using Firefox.
Does
Yes.
Julian
Brian J. Murrell wrote:
On Mon, 2008-05-05 at 16:36 -1000, Julian Yap wrote:
That was a bug in the release.
From the 1.4.20-rc1 Changelog:
2008-04-30 16:30 + [r114891] Russell Bryant [EMAIL PROTECTED]
So basically, r114891 was a fix to AST-2008-006? So if you applied
Tilghman Lesher wrote:
On Thursday 08 May 2008 11:03:34 John Novack wrote:
Tilghman Lesher wrote:
On Thursday 08 May 2008 00:52:35 Steve Totaro wrote:
I can certainly post more dirt from Mark's previous right hand man
if you wish to continue this argument.
I'd enjoy the chance to debunk the
For zap channels, you do have the c option on the dialgroup which
requires that you press # before the call is connected.
Works great for my mobile ;)
Julian
Atis Lezdins wrote:
On Sun, May 11, 2008 at 8:49 PM, Matt Watson [EMAIL PROTECTED] wrote:
I just took a quick look at the dialplan
Bryson Medlock wrote:
I'm trying to convince my employer to deploy an Asterisk based system, but
one member of the leadership team is against it. The rest of the team is
for it, but he's convinced them that we should find other organisations in
the Joplin, MO area who are using Asterisk first
I have a lot of recordings from asterisk in a .gsm format. I would like
to play these files from a web browser (IE, firefox and opera)
What do I need to do in order to achieve this goal ?
Thanks
Julian
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: May 15, 2008 7:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] playing .gsm sounds through a web browser
On Thursday 15 May 2008 18:26:15 Julian Lyndon-Smith wrote:
I have a lot of recordings from asterisk in a .gsm format. I would like
handle a wav
file.
On Thu, May 15, 2008 at 7:26 PM, Julian Lyndon-Smith [EMAIL PROTECTED]
wrote:
I have a lot of recordings from asterisk in a .gsm format. I would like
to play these files from a web browser (IE, firefox and opera)
What do I need to do in order to achieve this goal ?
Thanks
Philippe Sultan wrote:
Friends,
a new dialplan application is now available for testing :
http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/
Sounds very cool.
See below for more comments:
The corresponding feature request is located here :
Hi Philippe,
thanks for the replies. It all seems sensible.
Now, for a request ;)
How difficult would it be to have a JabberReceive Event *initiate* a
channel ?
This could be done by specifying a [EMAIL PROTECTED] in
jabber.conf
So, when a message is received by asterisk, a call is
I've got the following setup:
PhoneA -
router -
vpn -
router-
asterisk (SIP / ISDN)
PhoneB -
asterisk (SIP / ISDN)
If phone A is connected to phone B (sip-sip), there is a noticable delay
(up to 2-3 seconds) between me speaking and the other end hearing.
If phone A calls out
Hi Steve - the vpn is a consistent as the sip-IDSN has to go through
the VPN first to get to asterisk.
i.e. to make an outside call, PhoneA goes through the vpn to the
asterisk box, and out through isdn.
Julian
Steve Totaro wrote:
On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith [EMAIL
See below:
Erik Anderson wrote:
On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson
[EMAIL PROTECTED] wrote:
So now the PBX is over 1.2 Gig for the installation. Typical PBX
installs are under 600 Meg. This makes me wonder about server
stability, reliability and performance as uptime creeps on
This is what we use, with (seemingly) good success:
exten = answermachine,1,Answer
exten = answermachine,n,Wait(5)
exten = answermachine,n,WaitForSilence(1000,2)
Julian
Nicholas Blasgen wrote:
I'm trying to write an application for using after an agent has decided the
person on the other
If I am dialing a phone that has had a call forward put on, is there
anyway to stop asterisk following the call forward ?
I have a group of people that are on a ringall and one of them
forwarded their phone to an extension that answered and requested a
password. Anyone calling this group
Igor Zamocky wrote:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
attribute i
i: Asterisk will ignore any forwarding requests it may receive on this dial
attempt. (new in 1.4) Useful if you are ringing a group of people and one
person has set their phone to forwarded direct to voicemail
I am trying to track a strange bug down, and need to ask a really stupid
question, just so I can eliminate the possibility ..
When a SIP channel is hung up, I import a variable called MEETMEROOM
from the BRIDGEPEER channel, and if it is set, jump to another part of
the dialplan.
[snip]
exten
Hi Brent,
comments inline:
Brent Davidson wrote:
Julian Lyndon-Smith wrote:
I am trying to track a strange bug down, and need to ask a really stupid
question, just so I can eliminate the possibility ..
When a SIP channel is hung up, I import a variable called MEETMEROOM
from
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 so
much better than marketing blurb ;)
We
Hi Steve,
Steve Kennedy wrote:
On Wed, Oct 01, 2008 at 12:53:41PM +0100, Julian Lyndon-Smith wrote:
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
I seem to remember that there was a change to q931.c that meant a line
did not drop immedately, and then that change was reverted ?
I think that these are the lines of code:
/* wait for a RELEASE so that sufficient time has passed
for the inband audio to be heard */
if
Gordon Henderson wrote:
[snip]
That's a prety good price - I didn't think the dual-core Atoms were out
yet. I'll wait until I can get something in the UK..
Damn - I've just found it in thet UK too:
http://www.lambda-tek.com/componentshop/index.pl?prodID=1606310
Must
What options are available for installing an asterisk system onto a
bare-metal system ?
Ones that I have seen:
pbx-in-a-flash
trixbox
astlinux
What I am trying to achieve is to be able to shove a cd / usb into a
machine and have it install asterisk, complete with my .conf files.
I also need
show application RemoveQueueMember
-= Info about application 'RemoveQueueMember' =-
[Synopsis]
Dynamically removes queue members
[Description]
RemoveQueueMember(queuename[|interface[|options]]):
Dynamically removes interface to an existing queue
If the interface is NOT in the queue and
Dan Austin wrote:
Julian wrote:
show application RemoveQueueMember
-= Info about application 'RemoveQueueMember' =-
[Synopsis]
Dynamically removes queue members
[Description]
RemoveQueueMember(queuename[|interface[|options]]):
Dynamically removes interface to
I want to rerecord the 1 2 3 ... 0 sounds, but not overwrite the
defaults. So, I've recorded them into a custom directory
/var/lib/asterisk/sounds/custom
I was hoping to be able to do the following:
exten = foo,1,Set(CHANNEL(language)=custom)
exten = foo,2,SayDigits(1234567890)
however, I
I was wanting to automate entirely a blind transfer. We are not yet
using a powerdialler, so when we hit an answermachine we have to
manually leave a message.
In order to make this a little quicker, I want to leave a standard
message on the answermachine.
attempt #1. Use the blind transfer
Func_odbc is your friend.
Check out func_odbc.conf for odbc access from the dialplan
Check out res_odbc.conf to allow you to use odbc as a realtime source
Julian.
Callum McGillivray wrote:
I was hoping for something more along the lines of the Asterisk CMD
MySQL().
I could always resort to
That would be in your odbc.ini setup (normally /etc/odbc.ini)
Julian
Callum McGillivray wrote:
Ahhh - now this seems like the kind of thing that I was after
Where do I define the available DSN's though?
Julian Lyndon-Smith wrote:
Func_odbc is your friend.
Check out func_odbc.conf
in 1.4, func_odbc is your friend.
Julian.
David wrote:
Hello,
I'm looking to do the following, and I wonder if Asterisk can be used for it,
and if yes, if anyone can point me to the relevant information (commands,
sample config...):
1. Caller dials 111, 222 or 333.
2. Based on the dialed
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 ?
to all if I join the
second user with the q option.
Is there any way of playing a file *to* a meetme conference ? This way I
could play the sound to the first user before I join the second user to
the conference.
Julian.
-kn0x
On 5/24/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I am
Having scoured the web, I still am no better off ..
I have 2 Aastra480i's , and 120+ cisco 7940's :) . I am trying to decide
which model to use going forward when we purchase more kit. They both
seem much on a par regarding features.
Q1: Is there anyway of making the cisco auto-answer
We are having some major problems with app_swift since we went live. It
is regularly segfaulting.
I don't know if this is my fault or not, but here's the story:
Installed the cepstral voices (at the time, 4.0) on our test system
(2.6.9-42.0.10.ELsmp)
and later added some extra voices (now
4.2.0.
I doubt different voices behave differently, but just in case, I use the
$7 Damien voice.
Moj
Julian Lyndon-Smith wrote:
We are having some major problems with app_swift since we went live.
It is regularly segfaulting.
I don't know if this is my fault or not, but here's the story
See below:
Atis wrote:
Hi,
I'm currently migrating to 1.4 and have problems changing deprecated
AgentCallbackLogin to AddQueueMember.
I have dynamic queues and dynamic agents (MySQL Realtime), and
pseudo-dynamic agents.conf (with huge amount of possible agent
numbers).
Agent login is done
Is there anyway on knowing in the h extension if a call has been ended
as a result of a transfer ?
i.e.
1) A calls B.
2) B transfers A to C.
3) B gets hung up.
4) A talks to C
at (3) i need to know if this is a normal hangup (A or B has hung up) or
if it is a result of the transfer.
Julian.
'in use'.)
;
; ringinuse = no
Julian
Thanks,
James
James Fromm wrote:
NICE! That did the trick.
Thanks!
Julian Lyndon-Smith wrote:
try autopause in queues.conf
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asterisk-users mailing
We are in the process of implementing realtime voicemail. I was wanting
to stress-test the system to see if or when it would fall over.
Is it possible to use sipp to create say 250 calls, each of which leaves
a message in the voicemail ?
My dialplan is currently
[default]
exten =
voicemail messages (with 50 simultaneous calls) leaving
a 6-7 second voicemail (using .wav, .WAV and .gsm)
I'm *really* going to try and hurt it now ;)
Julian
Victor Toofic wrote:
El mar, ene 23 de 2007 a las 14:44 +, Julian Lyndon-Smith comentaba:
however, if I use sipp to test this, I get
BJ Weschke wrote:
On 2/2/07, Thomas Winter [EMAIL PROTECTED] wrote:
Hi,
I have an queue stored in relatime and defined members called through
LOCAL/
I found out that if I call the members through the LOCAL think the queue
statistics is not updated.
Any idea, or isnt possible to call
I've got around 45 people who need to place calls from our inhouse app.
What is the considered best practice for placing these calls:
1) All clients connect to astmanproxy, and use AMI API Originate command
2) All clients connect directly to the astersik AMI and use the API
Originate command
Kyle Sexton wrote:
Started playing with 1.4 and I'm curious what uses people have come up
with for the Jabber integration? So far I can think of presence based
call routing, but I'm sure there are other ideas. How are YOU using
the new Jabber features in 1.4? :)
We've been using it since
, and changed the
colours / labels / tooltips according to the presence message of each
jabber client.
Julian.
--
Chris Earle
Julian Lyndon-Smith [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Kyle Sexton wrote:
Started playing with 1.4 and I'm curious what uses people have come up
Given a choice, and a green-field site, would you
a) Have a separate network (switches etc) for your data and phone
b) Use the same network, but use VLAN's ??
What are the pro's and con's of each ?
TIA
Julian
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Gordon Henderson wrote:
On Wed, 28 Feb 2007, asterisk wrote:
What is the make of the existing pabx? Be aware that if it is an older
pabx
the signaling over the isdn30 could actually be dass2 which is not
compatable with asterisk cards, rather than true isdn30e.
This is another issue I
I've gotten hold of a Tesco Internet Phone which is a dect phone with
the base connecting to the pc via usb.
Has anyone been able to get this working with any softphone like xlite ?
It seems as if the tesco internet phone uses IAX - the software that
comes with it is a rebranded firefly (or
Of Julian Lyndon-Smith
Sent: Thursday, 1 March 2007 1:51 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Tesco Internet Phone
I've gotten hold of a Tesco Internet Phone which is a dect phone
with
the base connecting to the pc via usb.
Has anyone been able to get this working with any
Olle,
A couple of questions:
A) If it works for 1.2, I presume that it would work for 1.4 ?
B) We currently use realtime for voicemail. Is there a realtime engine
for Minivoicemail ?
Julian
Olle E Johansson wrote:
Just to tease users to test Mini-Voicemail:
*CLI show function
what's the easiest way of using cepstral voices with asterisk ? On their
website, in the ssml page
(http://www.cepstral.com/cgi-bin/support?page=ssml), they say
Asterisk PBX
SSML can be used with Cepstral voices in Asterisk by simply embedding
the markup into the input text.
what input text
Kai-Uwe Jensen wrote:
There's also an app_swift available at http://www.loopfree.net/app_swift/
Thanks to all that responded. I've used app_swift as mentioned above and
it suits my needs.
Thanks again
Julian
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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
Oh man - the second I send this, I find the answer.
say-as type=currency12345.44/say-as
Sorry for the waste of bandwidth.
Julian
Julian Lyndon-Smith wrote:
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
the minivm app itself, rather than the whole branch ?
TIA
Julian.
Olle E Johansson wrote:
11 mar 2007 kl. 10.04 skrev Julian Lyndon-Smith:
Olle,
A couple of questions:
A) If it works for 1.2, I presume that it would work for 1.4 ?
There is a branch called minivoicemail-1.4.
Modules for 1.2
We only present the 6 digits ... and they give us 6 digits. For our
outbound calls, for the the numbers 01702 1234[00-99] we have to present
1234[00-99].
BT isdn pri line.
Julian.
Steve Kennedy wrote:
Just a FYI to the list.
It seems that although BT only present 6 digits (as standard) for
at the moment, if our agents make a call and they get an answering
machine they have to wait for the beep before leaving a message.
I would like them to be able to transfer the call to an extension where
an automated message can be left as soon as they know it is an A/M.
However, how do I
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
Lyndon-Smith wrote:
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
Arrgh. this is driving me nuts. Can anyone put me out of my misery ?
Pretty please ;)
Julian Lyndon-Smith wrote:
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
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