Hi Enrico,
have you tried with busydetect=yes? It (sometimes) worked for me with
Asterisk 1.2.
Giorgio
Enrico Maistro wrote:
Hi,
I'm trying to get up and running a TDM400 with a standard italian pots
line but i'm having
problems at getting asterisk to detect when the line get answered
Hi Marino,
I tried to connect zoiper directly to the provider with the same account
parameters I'm using with Asterisk. Zoiper connects without problems. It
is true tnet.it is not resolvable but I can use the proxy URL
sip.tnet.it which seems to work with Zoiper but not with Asterisk. I'm
Hi Giorgio,
Just to recap:
1) you are able to connect to tnet.it by using the same account of your
asterisk box. There is no issue related to your account.
2) Could you please confirm that you are running zoiper from the same box
used by asterisk? If yes we can exclude some generic network
Hi Enrico.
In Italy the polarity reversal is never used.
I'm using the TDM400 with an FXO port in Italy with the config reported
below and is working properly in any situations:
--- zaptel.conf ---
fxsks=1
loadzone=it
defaultzone=it
--- zapata.conf ---
[channels]
language=en
context=from-tdm-fxo
Hi Marino,
1) yes I can connect using the account
2) no, I'm running zoiper on a different machine. I'm using an Asterisk
server which is not behind nat as for the machine zoiper is runnin' on.
The Asterisk server is directly connected to internet, I wanted to avoid
nat problems, that's why.
Thanks Noah.
It is now properly running. Thanks again
regards
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: Tuesday, July 15, 2008 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi Giorgio,
RE my point 2:
You should test a sip client, whatever you want, on your linux/asterisk box
just to double check that this box works fine.
If you are abel to connect with a sip client from tour asterisk box we will
be sure that the network configuration is ok.
You have no natt but
Check dns server entries in asterisk box . /etc/resolv.conf . Put
opendns servers ip there just to test . opendns ip's are
208.67.220.220 and 208.67.222.222
On Tue, Jul 15, 2008 at 2:19 PM, map [EMAIL PROTECTED] wrote:
Hi Giorgio,
RE my point 2:
You should test a sip client, whatever you
Hi Noah,
Hi Enrico -
I'm trying to get up and running a TDM400 with a standard italian pots
line but i'm having
problems at getting asterisk to detect when the line get answered on
outgoing calls.
I'm using asterisk 1.6 beta 9 with zaptel 1.4.11.
Zaptel channels use fxs_ks signalling
Hi Giorgio,
Giorgio Incantalupo wrote:
Hi Enrico,
have you tried with busydetect=yes? It (sometimes) worked for me with
Asterisk 1.2.
Giorgio
I'm already using busydetect=yes to detect hangup and busy conditions
with good results, but it doesn't seem to be of any help on detecting
Hi Marco,
Marco Signorini wrote:
Hi Enrico.
In Italy the polarity reversal is never used.
Good to know... at least i can stop messing with it.
I'm using the TDM400 with an FXO port in Italy with the config reported
below and is working properly in any situations:
--- zaptel.conf ---
Hi Enrico.
I'm quite sure that the differences you have in the zapata.conf doesn't
have any effect on the problem.
If I'm not wrong:
language=it tells asterisk to use the italian sounds (if available)
for any calls related to this zap channel;
rxgain = 0.0 is related only to perceived audio gain
On Mon, Jul 14, 2008 at 08:56:40PM +0200, Enrico Maistro wrote:
Hi,
I'm trying to get up and running a TDM400 with a standard italian pots
line but i'm having
problems at getting asterisk to detect when the line get answered on
outgoing calls.
AFAIK chan_zap can only detect answer if it
Hi all,
I solved it
I tried with an Asterisk 1.4 test box.
It said:
ast_get_srv: SRV lookup for '_sip._udp.tnet.it' mapped to host
sip.tnet.it, port 5060
and...it seems to work!!
So I put srvlookup=yes on Asterisk 1.2 and IT WORKS!!!
Now I try to make calls.
Thank you all for
On Sun, 2008-07-13 at 10:22 -0400, Brian J. Murrell wrote:
I have a wildcard 100 xp on my pots line and all was working just fine
up until a few days ago when all of a sudden it stopped receiving caller
id on incoming calls. I know caller id is being presented on the line
as the analog set on
Brian J. Murrell wrote:
One thing I have noticed is that in the cases where the wildcard cannot
determine the CID (i.e. because the rxgain is up around 10.5), I get
this in my asterisk console:
[Jul 15 08:04:09] NOTICE[26696]: chan_zap.c:6670 ss_thread: Got event 18
(Ring Begin)...
And
Hi guys,
Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server?
The pbx doesn't have sip and I want to come in off of a sip trunk and
interface with the older system.
I know I can use a pri card to hook in to the phone network, but can I use
this same card to feed back the
On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote:
After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri,
zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
working.
THis isn;t going to fix your problem... but just FYI, you don't need to
install
Hi Tzafrir,
you're right. I think I've completely misunderstood the problem.
If the problem is that asterisk is not able to write in the CDR the
proper line answer status, I can confirm that even my installations
behave the same.
Sorry Enrico for my fault and thank you to Tzafrir for the
I cannot tell for sure for any system, but we have an old Portmaster PM3
hooked-up from one port of our Sangoma A104d card, another one being from
telco.
So, yes you can emulate the telco from a sangoma A10x card. Here's what I
have in my zapata.conf :
;Sangoma A104 port 1 [slot:12 bus:0
On Tue, 2008-07-15 at 22:31 +1000, Rob Hillis wrote:
Brian J. Murrell wrote:
One thing I have noticed is that in the cases where the wildcard cannot
determine the CID (i.e. because the rxgain is up around 10.5), I get
this in my asterisk console:
[Jul 15 08:04:09] NOTICE[26696]:
Actually what I'm doing is interfacing the legacy pbx and converting it to
use sip for its way out to the world.
The phone vender I'm working with says his system requires b8zs signaling
and uses the esf frame type.
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
The configuration for a PM3 would be the same for a PBX. One additional
note, put the channels on the PBX PRI in its own context, and then set
that context up in your dialplan to forward the calls out to your SIP
provider.
-Original Message-
From: [EMAIL PROTECTED]
Hi All,
When I use re-invite, does the Asterisk server stay in the SIP
conversation, and just RTP traffic diverts, or does the SIP transfer
away from the A*k server too ?
Thanks,
Adrian
___
-- Bandwidth and Colocation Provided by
Hi,
I'm getting this bizarre problem. Whenever I dial (through misdn) and
try to listen to my music on hold, I get this:
-- Started music on hold, class 'default', on channel 'mISDN/3-u72'
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested
On Fri, Jul 11, 2008 at 03:59:22PM -0500, Joe Greco wrote:
On Fri, Jul 11, 2008 at 12:58:59PM -0700, Daniel Hazelbaker wrote:
Really? You have an RJ-21X block that contains both analog AND T1
wires? That's really uncommon. I hope they at least put the red
special service caps on
Vazquez David wrote:
Hi,
I'm getting this bizarre problem. Whenever I dial (through misdn) and
try to listen to my music on hold, I get this:
-- Started music on hold, class 'default', on channel 'mISDN/3-u72'
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing
I am using realtime on two boxes, one running 1.4.10.1 and one running
1.4.11. Everything works fine except for when I make a database change,
such as a phones password. I change the DB, I prune the peer, I see it
is gone and then I see it show up again in sip show peer , but
everything
One thing I have noticed is that in the cases where the wildcard cannot
determine the CID (i.e. because the rxgain is up around 10.5), I get
this in my asterisk console:
[Jul 15 08:04:09] NOTICE[26696]: chan_zap.c:6670 ss_thread: Got event 18
(Ring Begin)...
It is odd that it would
Hi Adrian -
When I use re-invite, does the Asterisk server stay in the SIP conversation,
and just RTP traffic diverts, or does the SIP transfer away from the A*k
server too ?
I'm sure somebody will correct me if this is wrong, but I believe the
signalling must stay with asterisk, as asterisk
On Tue, Jul 15, 2008 at 12:05 PM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
I am using realtime on two boxes, one running 1.4.10.1 and one running
1.4.11. Everything works fine except for when I make a database change,
such as a phones password. I change the DB, I prune the peer, I see
On Tue, 2008-07-15 at 12:49 -0400, Noah Miller wrote:
It is odd that it would work one day and not the next.
Indeed.
I'd have to
say, though that I've seen that rxgain/txgain values beyond +-8 seem
to yield unpredictable results in many areas,
Yeah, I was pretty alarmed months ago when I
Need to have a different TONE for any internal call (EXT OR TRANSFER) from
an external (outside) call.
Any suggestions?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix,
Hello All,
I am looking to find a way to provide international toll free access to
our Knoxville, TN (USA) office from our customers in the UK and in
Australia, and when I talked with ATT I was surprised to find out how
expensive they are... Surely, other businesses are not paying this much -
Larry,
Give us a call (646) 862-1555
/jon
- Original Message -
From: Larry Costigan
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, July 15, 2008 2:22 PM
Subject: [asterisk-users] Toll Free International Number
Hello All,
I am looking to
Folks:
Does anyone know of a replacement for meetme that provides native G729
support? The transcoding back and forth from/to 711 is eating too much
processor for what we're doing...
Many thanks,
--ag
--
Artie Gold
F4W, Inc.
___
-- Bandwidth and
Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia:
Need to have a different TONE for any internal call (EXT OR TRANSFER)
from an external (outside) call.
Any suggestions?
Fidel,
I do not know what kind of tone you mean:
The sound of a phone that signals a call coming from
This one!
The sound of a phone that signals a call coming from internal/external
My phones are SIP, I do not know what ZAP means or what it does.
Thanks for your reply!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin
Hoffmeister
Sent:
Internal and external calls can be distinguished generally by the phone
number. A prefix or the number of digits of the phone number. For example,
you could use a digit prefix followed by a interval of time to call a
internal number.
Examples:
Internal number: 0,1234
External number:
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
On Tue, Jul 15, 2008 at 2:37 PM, Fidel Garcia [EMAIL PROTECTED]
wrote:
This one!
The sound of a phone that signals a call coming from internal/external
My phones are SIP, I do not know what ZAP means or what it does.
Thanks for your
Hi,
I need libpri, because I have a TE110P E1 with a PRI ISDN service.
2008/7/15 Matt Watson [EMAIL PROTECTED]:
On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote:
After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading
libpri,
zaptel, the incoming calls to a TDM400P REV I,
Thank you,
yes, I changed the PCI Slot and it's the same,
I get a used card from a customer with 2 FXO, same REV, that board was
working on the customer server, put it on mine, and stop working.
I put my board on his server and the board is working perfectly.
I had not test outgoing calls on
On Tuesday 15 July 2008 13:32:12 Artie Gold wrote:
Does anyone know of a replacement for meetme that provides native G729
support? The transcoding back and forth from/to 711 is eating too much
processor for what we're doing...
Buy a hardware transcoder board. There is simply no way to mix
That makes sense -- thanks!
--ag
On Tue, Jul 15, 2008 at 1:59 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 15 July 2008 13:32:12 Artie Gold wrote:
Does anyone know of a replacement for meetme that provides native G729
support? The transcoding back and forth from/to 711 is eating
On Jul 11, 2008, at 12:58 PM, Daniel Hazelbaker wrote:
I may have figured out the problem this morning, but I won't be able
to test for a few days (again, aggravating that the only T1 line I
have to test with is the live one). I noticed this morning while
telneted into the Adtran that when I
OK, I guess I need to show my ignorance -- what is the difference
between ulaw and signed linear?
on Tuesday 07/15/2008 Tilghman Lesher([EMAIL PROTECTED]) wrote
On Tuesday 15 July 2008 13:32:12 Artie Gold wrote:
Does anyone know of a replacement for meetme that provides native G729
Depending upon what cities you need, there are a lot of companies
offering this. I like IdeaSIP.com who have shown excellent call
quality and value over the years I've been using them.
/r
Can someone in this good group please help me with some advice as to who can
provide affordable and
Hi Uros -
I have problem using Asterisk.I have isdn-pri and openvox d110p card in my
computer.They are connected with RJ-45 (1,2,4,5 pins to the card and all
pins to the isdn done by telco workers). I got green led on isdn which is
sign that isdn is working and that is connected to openvox,
It depends on which type of SIP device you have that determines on how
you signal a distinctive ring. You need to change the SIP Header like:
exten = s,n,SIPAddHeader(Alert-Info:Bellcore-r8)
where the number after the 'r' signifies a different ring tone but some
devices uses different names
Hi,
How can I be notified anytime a given warning message appears in Asterisk
logs ?
I've got a running system that produces cause 34 warnings (Unable to
create channel of type 'Zap' (cause 34 - Circuit/channel congestion)) once
or twice a week.
I would like to like to be notified (by email,
On Tuesday 15 July 2008 14:24:30 John covici wrote:
on Tuesday 07/15/2008 Tilghman Lesher([EMAIL PROTECTED])
wrote
On Tuesday 15 July 2008 13:32:12 Artie Gold wrote:
Does anyone know of a replacement for meetme that provides native G729
support? The transcoding back and forth from/to
On Jul 11, 2008, at 12:28 PM, Robert McNaught wrote:
Has anyone deployed a hosted environment like enswitch using EC2? I
was wondering if anyone had any thoughts on concerns on the
feasibility in doing this using cloud computing?
For setting up a VoIP service provider and not having the
I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using
CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I
try to build Asterisk this is the error I'm getting.
src/add.c:1: error: CPU you selected does not support x86-64 instruction set
I
All:
Has anyone else on the list had any experience with the new Adtran IP712
phones? I have taken the stock config file and been able to get simple
registrations and basic call processing to work properly, however, I'm
finding little to no documentation on how to configure advanced options
perl script.
Olivier wrote:
Hi,
How can I be notified anytime a given warning message appears in
Asterisk logs ?
I've got a running system that produces cause 34 warnings (Unable
to create channel of type 'Zap' (cause 34 - Circuit/channel
congestion)) once or twice a week.
I would
Hi -
I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm
using CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk.
When I try to build Asterisk this is the error I'm getting.
src/add.c:1: error: CPU you selected does not support x86-64 instruction set
HI all,
I am having issues with the gui on my AA50.
under Voice Menus Add new Step Go to Time based rule.
It allows me to select Go to Time based rule from the menu but no options
come up when selected.
I've tried all browsers but no luck.
Thanks
David.
voxbone.com
- Original Message -
From: Larry Costigan
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Toll Free International Number
Date: Tue, 15 Jul 2008 14:22:49 -0400
Hello All, I am looking to find a way to provide international toll
I am new to asterisk, and I am having some troubles.
I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and
asterisk-gui installed on centos (I built everything using ./configure,
make, make install, make samples). I connected to the GUI interface and
created two new users. I
Hi John -
I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and
asterisk-gui installed on centos (I built everything using ./configure,
make, make install, make samples). I connected to the GUI interface and
created two new users. I used the two users accounts to connect up
Are your phones behind NAT?
This should be an issue with rtp port communication.
Gerard.
--Original Message--
From: John Koenig
Sender: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Jul 15, 2008 6:47 PM
My internal calls start in an entirely different context than calls
coming in externally. There's never any confusion about where the call
is coming from and I don't use prefixes.
Allann Jones wrote:
Internal and external calls can be distinguished generally by the
phone number. A prefix or
Have you upgraded to the latest version?
We found a few bugs went away on our test unit when we did that.
PaulH
Sydney Web Hosting wrote:
HI all,
I am having issues with the gui on my AA50.
under Voice Menus Add new Step Go to Time based rule.
It allows me to select “Go to Time based
I had an issue where I put a comma in the prepend digits string pn
call plans and then the call plan menu would no longer load.
It parses the menu from the text file so I used the file editor to
clear the offending line and my menu came back. Not sure if thats your
issue but I was surprised
I had issues like this on one installation that cleared up when I turned
ACPI and APIC?? off in bios.
Darren Wiebe
[EMAIL PROTECTED]
Gerard A. Matthew wrote:
Are your phones behind NAT?
This should be an issue with rtp port communication.
Gerard.
--Original Message--
From: John
That could be...I only have ports 5060 and 8088 open on the firewall.
Should another port be open?
The phone I am using are pstn phones connected to a 2 port Linksys PAP2.
I made sure that I checked the NAT option under the user account and
enabled NAT Keep Alive under the PAP2 management
Time based rules are no longer in use. Contact Digium support that you
got received with your aa50.
-bk
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Lists wrote:
Hi all,
I am wanting to change the sound files from the standard ones to a New
Zealand voice pack.
I have copied the files into the /var/lib/asterisk/sounds directory and
chowned them to asterisk:asterisk and chmod 420 to match
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Vinícius Fontes wrote:
As RTP packets have a sequential number, is there some logging/debugging
option in Asterisk to monitor how many packets have been lost on a SIP call?
You could use rtcp stats if the endpoints support it.
- --
Kind Regards,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John Faubion wrote:
Try dropping the IAX2 and only use SIP. Don't ask why?
Well in our case we were NOT using IAX at all. Strictly SIP.
You could be hitting an overloaded router or whatever along
the way, 10mbs fiber does not mean low
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Artie Gold wrote:
Folks:
This is my first post, so please let me know if I transgress in any way...
In updating to 1.4.21 recently, we've encountered a problem, when running
over a satellite connection (where the latency is considerable; a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steve Underwood wrote:
Dave Cotton wrote:
Joseph wrote:
On 07/11/08 18:37, Dave Cotton wrote:
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas Garstang wrote:
Thanks, but how does that extend the core functionality of Dial()? If Dial()
can't do it, how does a wrapper do it?
Did you see the patch that someone pointed out in your last
conversation? That does exactly that.
If you
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Loic Didelot wrote:
Hello,
I just got my Xorcom BRI bank. Seems to work. But I have some questions.
Is anyone getting good values using zttest?
Is it plugged into the BRI?
Is it the sync master?
i.e. xpp_sync
- --
Kind Regards,
Matt Riddell
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Noah Miller wrote:
Hi Leotis -
Now that you mention that, i didnt even know there was a gsm bug. I am using
asterisk 1.4.21.1, i visited the link you gave. I am guessing i will have to
patch my asterisk installation, i am reading, the bug report
List,
We're working on an upcoming job that may require us to access a web
service (WS). I'm curious to hear peoples thoughts on the best way to
do this with asterisk. We'll be submitting a single number to the WS
and it will return a success or error.
One solution would be to write a simple
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason Dixon wrote:
On Tue, Jul 08, 2008 at 11:00:43AM -0400, Jason Dixon wrote:
On Tue, Jul 08, 2008 at 12:10:05PM +1200, Matt Riddell wrote:
Action: Command
Command: show queue my_queue_name
ActionID: my_queue_name_12345
This does not appear to
On Jul 15, 2008, at 10:08 PM, Paul Belanger wrote:
List,
We're working on an upcoming job that may require us to access a web
service (WS). I'm curious to hear peoples thoughts on the best way to
do this with asterisk. We'll be submitting a single number to the WS
and it will return a
Try Adhersion and or Telegraph
-E
http://mobiquity.ws
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To
On Tue, 15 Jul 2008, Paul Belanger wrote:
We're working on an upcoming job that may require us to access a web
service (WS). I'm curious to hear peoples thoughts on the best way to
do this with asterisk. We'll be submitting a single number to the WS
and it will return a success or error.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steve Totaro wrote:
On Thu, Jul 10, 2008 at 11:43 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Thu, Jul 10, 2008 at 10:24 AM, Steve Underwood [EMAIL PROTECTED]
wrote:
Vinícius Fontes wrote:
When people release software under the GPL
On Jul 15, 2008, at 10:20 PM, Steve Edwards wrote:
curl() doesn't fire up another process. The response is returned as
just
one big chunk. In my case, it was the HTML to an entire web page :)
If you need to do a bunch of parsing, maybe an AGI calling libcurl --
saving a bunch of ugly
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Alexander Lopez wrote:
Snip
On Wed, Jul 9, 2008 at 10:50 AM, C F [EMAIL PROTECTED] wrote:
Very interesting article. I guess we won't know much more for another
few weeks:
http://www.breitbart.com/article.php?id=080709124916.zxdxcmkxshow_artic
I'm attempting to get Asterisk to talk with a VConsole ISDN simulator
that supports the following CAS protocols:
CAS EM Wink Start FGD
CAS EM Wink Start FGB
I've tried configuring the Asterisk end with em_w, featb, featd, featdmf
but with each of these, it either doesn't work at all, or I see
sip
Thanks,
--ag
On Tue, Jul 15, 2008 at 8:39 PM, Matt Riddell [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Artie Gold wrote:
Folks:
This is my first post, so please let me know if I transgress in any
way...
In updating to 1.4.21 recently, we've
Does anyone know of a bandwidth test that tests the upload with the download?
All of the ones I can find will test the upload then the download.
I from experience I have found that a 3M/768K DSL can only do about
256K/256K simultaneously.
The only way I have of testing it is with FTP uploads
Hi All,
I have got my voice menus setup. open hours and after hours.
What do I have to code in the main menu to do the following.
If between the hours of 9am - 5pm go to open hours
All other hours go to after hours
I've read all of the docs but don't quite understand it?
Cheers
David.
What do I have to code in the main menu to do the following.
If between the hours of 9am - 5pm go to open hours
All other hours go to after hours
You can do something like:
exten = main switch
no,n,GotoIfTime(9:00-17:00,mon-fri,*,*?rcl_off_opn)
___
Brian J. Murrell wrote:
Unless you want to invest in a
better card, you may just have to live with the problem.
Which means what, a multiport and multi-hundreds of dollar card? I'm
just a home user. I don't have hundreds of dollars to spend on a single
piece of phone hardware.
OK Ive done this.
exten=7000,n,GotoIfTime(9:00-17:00,mon-fri,*,*?rcl_off_opn)
7000 is the extension of main menu
Where do I put the reference to open hours menu in the statement above.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee, John
(Sydney)
90 matches
Mail list logo
