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 myths that you've
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
Queue(console,r)
would do what you want, but so you would need to have two entry points
to
queue.
Thanks Atis. Your suggestion did magic!
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asterisk-users mailing list
To
dmesg | grep -i zap
Should give you a version, and an echo cancellation technology.
PaulH
On Thu, 2008-05-08 at 17:05 +1000, Lee, John (Sydney) wrote:
The only things I set in relation to echo cancellation is in zapata.conf
where I put echocancel=yes
Ouch...any idea what echo
dmesg | grep -i zap
Should give you a version, and an echo cancellation technology.
Thanks Paul.
# dmesg | grep -i zap
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.6
Zaptel Echo Canceller: MG2
Zaptel Transcoder support loaded
Which is reasonably new, but an upgrade to the latest version (1.4.10.1)
will only take 5 minutes and is worth a shot.
PaulH
On Fri, 2008-05-09 at 15:24 +1000, Lee, John (Sydney) wrote:
dmesg | grep -i zap
Should give you a version, and an echo cancellation technology.
Thanks Paul.
@lists.digium.com
Subject: [asterisk-users] Newbie alert: VoIP hardware
Hello,
Please forgive me for i'm not an asterisk user yet. I've done as much
research as I can .. and have the following questions.
I'm setting up a new office and a home office and i'm shopping for hardware.
Office: 2 analog lines
Steve Repo wrote:
Hello,
Please forgive me for i'm not an asterisk user yet. I've done as much
research as I can .. and have the following questions.
I'm setting up a new office and a home office and i'm shopping for
hardware.
Office: 2 analog lines
Hardware: TDM412B (2 FXO, 1FXO)
Alan Lord wrote:
If you only have one analogue line why not just get a simple x100p card?
When you use OSLEC with them they work great here in the UK. I bought my
card from a USA based eBay seller. Total cost for card and shipping was
about £17.00
Respectfully, I don't agree. I've
Marco wrote:
Respectfully, I don't agree. I've purchased an original clone :-P of
the X100P card, on the long period they almost always have some
drawbacks... Faxing have been troubling for me. Don't know if it was for
the line or else, but with a Digium card I had no problem at all.
No
On Wed, 07 May 2008 09:58:04 +0100, Alan Lord wrote:
Marco wrote:
Respectfully, I don't agree. I've purchased an original clone :-P of
the X100P card, on the long period they almost always have some
drawbacks... Faxing have been troubling for me. Don't know if it was for
the line or
On Wed, May 7, 2008 at 8:17 AM, Michael Graves [EMAIL PROTECTED] wrote:
On Wed, 07 May 2008 09:58:04 +0100, Alan Lord wrote:
Marco wrote:
Respectfully, I don't agree. I've purchased an original clone :-P of
the X100P card, on the long period they almost always have some
On Wed, May 7, 2008 at 9:00 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Wed, May 7, 2008 at 8:17 AM, Michael Graves [EMAIL PROTECTED] wrote:
On Wed, 07 May 2008 09:58:04 +0100, Alan Lord wrote:
Marco wrote:
Respectfully, I don't agree. I've purchased an original clone :-P of
On Wednesday 07 May 2008 08:00:17 Steve Totaro wrote:
If your budget is tight and you want a decent card (not an X100P) with
room to upgrade, then check out
http://www.openvox.com.cn/products.php?genre_id=25 or
http://store.getvoicecards.com/index.php?cPath=66 they are the
reference design
On Wed, May 7, 2008 at 9:40 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 07 May 2008 08:00:17 Steve Totaro wrote:
If your budget is tight and you want a decent card (not an X100P) with
room to upgrade, then check out
http://www.openvox.com.cn/products.php?genre_id=25 or
On Wed, May 7, 2008 at 9:40 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Note that purchasing Digium boards helps pay for full time Asterisk
development, and purchasing clone boards does not pay for even a part-time
Asterisk developer.
--
Tilghman
BTW, I am all for having payed
On Wed, May 7, 2008 at 10:22 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Wed, May 7, 2008 at 9:40 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Note that purchasing Digium boards helps pay for full time Asterisk
development, and purchasing clone boards does not pay for even a part-time
On Wednesday 07 May 2008 09:40:21 Steve Totaro wrote:
Interesting results in Google for TDM400P TigerJet reference design.
http://www.google.com/search?hl=ensafe=offclient=firefox-arls=org.mozill
a:en-US:officialhs=h9Ppwst=1sa=Xoi=spellresnum=1ct=resultcd=1q=Tiger
On Wed, May 7, 2008 at 11:38 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 07 May 2008 09:40:21 Steve Totaro wrote:
Interesting results in Google for TDM400P TigerJet reference design.
http://www.google.com/search?hl=ensafe=offclient=firefox-arls=org.mozill
Besides the Background() app mentioned, you might like the WaitExten()
app
Thanks guys for your response.
I have had much success with Read() as below so that whenever I press a
key before the sound file finishes playing, it will read the digit and
move to the next line.
exten =
On Wednesday 07 May 2008 21:56:54 Steve Totaro wrote:
On Wed, May 7, 2008 at 11:38 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 07 May 2008 09:40:21 Steve Totaro wrote:
Interesting results in Google for TDM400P TigerJet reference design.
If your budget is tight and you want a decent card (not an X100P) with
room to upgrade, then check out
http://www.openvox.com.cn/products.php?genre_id=25 or
http://store.getvoicecards.com/index.php?cPath=66 they are the
reference design that Digium used on previous cards and are very well
the relaxdmtf (or similar) option in zaptel can make this work a bit
better...but it's a try at your own risk option!
PaulH
Thanks Paul.
I have further findings into the problem.
While the message is being played, if I press a key during the pause
or break between words, then the key will
Ouch...any idea what echo cancellation your system is using?
PaulH
On Thu, 2008-05-08 at 14:55 +1000, Lee, John (Sydney) wrote:
the relaxdmtf (or similar) option in zaptel can make this work a bit
better...but it's a try at your own risk option!
PaulH
Thanks Paul.
I have further
On Thu, May 8, 2008 at 12:46 AM, Steve Repo [EMAIL PROTECTED] wrote:
If your budget is tight and you want a decent card (not an X100P) with
room to upgrade, then check out
http://www.openvox.com.cn/products.php?genre_id=25 or
http://store.getvoicecards.com/index.php?cPath=66 they
Lee, John (Sydney) wrote:
the relaxdmtf (or similar) option in zaptel can make this work a bit
better...but it's a try at your own risk option!
PaulH
Thanks Paul.
I have further findings into the problem.
While the message is being played, if I press a key during the pause
or break
On Thu, May 8, 2008 at 12:27 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 07 May 2008 21:56:54 Steve Totaro wrote:
On Wed, May 7, 2008 at 11:38 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 07 May 2008 09:40:21 Steve Totaro wrote:
Interesting results in
Hello,
Please forgive me for i'm not an asterisk user yet. I've done as much
research as I can .. and have the following questions.
I'm setting up a new office and a home office and i'm shopping for hardware.
Office: 2 analog lines
Hardware: TDM412B (2 FXO, 1FXO)
Link:
If you have access to the console you can do many things.
For instance, you can originate test calls.
Tzafrir, thanks for your response and sorry for not being specific.
You raised a very good point about accessing the console.
For example, I made a major change to some of the config files
On Fri, May 02, 2008 at 05:39:40PM +1000, Lee, John (Sydney) wrote:
If you have access to the console you can do many things.
For instance, you can originate test calls.
Tzafrir, thanks for your response and sorry for not being specific.
You raised a very good point about accessing the
I will be installing Asterisk in a few offices which I don't have any
colleagues over there to help me.
Let's suppose I installed Asterisk in such a site. I tested it to my
satisfaction and I went back to my home office.
One day, a customer called me to say that he had a problem calling out
or
On 1 May 2008, at 08:17, Lee, John (Sydney) wrote:
I will be installing Asterisk in a few offices which I don't have any
colleagues over there to help me.
Let's suppose I installed Asterisk in such a site. I tested it to my
satisfaction and I went back to my home office.
One day, a customer
On Thu, May 01, 2008 at 05:17:17PM +1000, Lee, John (Sydney) wrote:
I will be installing Asterisk in a few offices which I don't have any
colleagues over there to help me.
Let's suppose I installed Asterisk in such a site. I tested it to my
satisfaction and I went back to my home office.
One
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 =
Lee, John (Sydney) wrote:
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} =
I am trying to find out if Polycom (I am using IP601) can display the
speed dial list using last name first instead of first name first.
Currently, the speed dial list displays first name first.
Thanks.
___
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Just want to know if anyone has used instant messaging using Polycom and
Asterisk.
From Google, I did not really see IM being mentioned at all. It appears
no one is interested to implement it in Asterisk. Or I guess people
would rather use Jabber or other IM messengers.
DND does not do anything for me BLF-wise either (shame). Simply
picking up
the handset won't do, at that point the phone is giving you a dialtone
but
nothing is sent to the server. You actually have dial out. Try
actually
calling somebody, the state should change to InUse.
Thanks Mike and
I believe this isn't a Polycom thing, but the nature of SIP devices in
general. But, that said, Polycom should start making IAX desk phones. :-)
- Chris
Lee, John (Sydney) wrote:
DND does not do anything for me BLF-wise either (shame). Simply
picking up
the handset won't do, at
I am working on Polycom IP601 console with expansion module.
I want to put on the BLF (busy lamp field) feature on all the
contact/speed dial names I put on the console but I could not get it to
work.
*CLI core show version
Asterisk 1.4.13 built by root @ hostname on a i686 running Linux on
I am working on Polycom IP601 console with expansion module.
I want to put on the BLF (busy lamp field) feature on all the
contact/speed dial names I put on the console but I could not
get it to work.
John,
If I understand correctly (and that's my experience) the BLF will only light
up
If I understand correctly (and that's my experience) the BLF will only
light
up when the phone is ringing/on a call. Asterisk doesn't support all
those
fancy status that you can select from the phone.
Mike, thanks for your response.
I think my test is worse than that. I pressed DND on one
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lee, John (Sydney)
Sent: Friday, April 18, 2008 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie Polycom: Subscription/Presence
Problem
If I understand correctly (and that's my experience
config looks good.
Regards,
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Lee, John (Sydney)
Sent: Friday, April 18, 2008 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie Polycom
I am exploring the contacts directory in Polycom and I am wondering if a
big -directory.xml on the boot server will eat up the memory
and crash the Polycom phone once downloaded onto the phone.
The asterisk directory extension is good but because users cannot see
the names I thought to
If you are busy doing something else and you hear this soft, pleasant
and unobtrusive sound, you suddenly realize Oh sh..., a Polycom
just rebooted! :)
___
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asterisk-users mailing
That would at least be long enough to cover the entire boot process. ;)
Lee, John (Sydney) wrote:
It's played at the completion of the boot process. It's always been
very quiet on the models I've worked with.
Thanks Erik. I can probably replace it with my beloved Mozart Symphony
no 40
On Tue, Apr 08, 2008 at 12:25:09AM -0500, Erik Anderson wrote:
On Tue, Apr 8, 2008 at 12:06 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
When I downloaded the sip and bootrom from Polycom website, I noticed a
file called SoundPointIPWelcome.wav. However, I have no idea where and
when
] On Behalf Of Rob Hillis
Sent: Tuesday, 8 April 2008 6:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie Polycom: Where
isSoundPointIPWelcome.wav used?
That would at least be long enough to cover the entire boot process. ;)
Lee, John (Sydney
Any suggestion for a headset (cord and cordless) for IP601?
Any good (and economical) ones from Polycom or Platronics?
I don't know about cordless, but for corded, I've had great success
with Plantronics H91N's.
- Noah
___
-- Bandwidth and
I would recommend the Plantronics CS70N
On Mon, Apr 7, 2008 at 11:47 AM, Noah Miller [EMAIL PROTECTED] wrote:
Any suggestion for a headset (cord and cordless) for IP601?
Any good (and economical) ones from Polycom or Platronics?
I don't know about cordless, but for corded, I've had great
When I downloaded the sip and bootrom from Polycom website, I noticed a
file called SoundPointIPWelcome.wav. However, I have no idea where and
when it was used. I played the wav file but I have never heard the
phone using this wav file before. Does anyone know what it is used for?
On Tue, Apr 8, 2008 at 12:06 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
When I downloaded the sip and bootrom from Polycom website, I noticed a
file called SoundPointIPWelcome.wav. However, I have no idea where and
when it was used. I played the wav file but I have never heard the
It's played at the completion of the boot process. It's always been
very quiet on the models I've worked with.
Thanks Erik. I can probably replace it with my beloved Mozart Symphony
no 40 :-)
___
-- Bandwidth and Colocation Provided by
Any suggestion for a headset (cord and cordless) for IP601?
Any good (and economical) ones from Polycom or Platronics?
Thanks.
___
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asterisk-users mailing list
To UNSUBSCRIBE or
SIP response 486 Busy Here is returned unless a divert contact is set
up in the phone config. I did a search through the SIP 3.0 Admin Guide
and didn't see any way of returning a different SIP response.
All Polycom phones use the same firmware and bootroms - one reason why
the sip.ld is so damn large for them.
Thanks Rob.
Alleluia! Rob, I will take your word for it - it solves all my worries
in deploying different models to the same environment like IP5XX and
IP6XX.
Can't you just use the same bootrom for all your polycom phones?
PaulH
On Fri, 2008-03-28 at 15:38 +1100, Lee, John (Sydney) wrote:
I have a question about DHCP and boot server supporting more than 1
model of Polycom phones.
According to Polycom standards, Polycom phone boots up to get a
On Fri, Mar 28, 2008 at 12:05 AM, Paul Hales [EMAIL PROTECTED] wrote:
Can't you just use the same bootrom for all your polycom phones?
PaulH
On Fri, 2008-03-28 at 15:38 +1100, Lee, John (Sydney) wrote:
I have a question about DHCP and boot server supporting more than 1
model of
Paul Hales wrote:
Can't you just use the same bootrom for all your polycom phones?
To elaborate in case it isn't obvious from above: Even if you needed
different config files or even SIP applications by phone, you don't have
to go to separate DHCP entries by phone. The MACADDESS.cfg file points
There is a sip.cfg entry divert.dnd.x.contact that is supposed to be
where the call goes if DND is enabled. You could presumably set that to
* plus the extention to go to the extension's voicemail, or to some
other dialplan to play whatever you want, though I haven't tried it.
Lee, John
What is your extensions.conf setup? that has alot to do with it (I
strongly suggest you use macros.) What SIP NNN code does the phone
return when DND?
On Mon, Mar 17, 2008 at 2:00 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
I am using Polycom IP600 phone. If I call a phone which has DND (do
I have a question about DHCP and boot server supporting more than 1
model of Polycom phones.
According to Polycom standards, Polycom phone boots up to get a DHCP
address and at the same time getting a boot server string (with username
and password) to logon to boot server to download SIP, bootROM
For this, I would recommend using a smart DHCP device, which supports
the passing of 'option 66' - for example, the edgemarc series of
routers.
With that, you could pass ftp://user1:[EMAIL PROTECTED] via dhcp
in order to provision the phone, and different credentials if you are
concerned about
All Polycom phones use the same firmware and bootroms - one reason why
the sip.ld is so damn large for them.
Lee, John (Sydney) wrote:
I have a question about DHCP and boot server supporting more than 1
model of Polycom phones.
According to Polycom standards, Polycom phone boots up to get a
Lee, John (Sydney) wrote:
I am working on a menu to accept input from a caller like as follows:
Exten = 100,1,Answer()
Exten = 100,n,Playback(LONG-MESSAGE)
Exten = 100,n,Read(OPTION,,2)
...
When I tested it, I noticed if I start pressing a key before the
Playback() is finished, the input
Hi,
I can get the message recorded and played correctly with wengo, but not with
zoiper. Is there any codec setting that I should fixed and how to fixed it?
On Fri, Mar 21, 2008 at 9:26 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Probably a codec issue. SIP debug while making a call would be
Hi,
I switched to Wengo and solved the one beatproblem. However, I am still
not able to listen to the recorded .wav sound. Can anyone please point me
to the right direction? How to listen to the .wav sound?
Thanks,
Pete
On Fri, Mar 21, 2008 at 9:34 AM, Carlos Rojas [EMAIL PROTECTED] wrote:
Probably a codec issue. SIP debug while making a call would be helpful.
Thanks,
Steve Totaro
On Fri, Mar 21, 2008 at 4:06 AM, Pete Kay [EMAIL PROTECTED] wrote:
Hi,
I switched to Wengo and solved the one beatproblem. However, I am still
not able to listen to the recorded .wav sound. Can
On Thu, Mar 20, 2008 at 01:27:47AM -0400, Al Baker wrote:
From a lot of experience - you are not being anywhere near paranoid
enough !!
Think dual RAID controllers, Dual power supplies off of, at a Minimum,
separate isolated circuits, with Hefty UPS that is in-line so it filters
In article [EMAIL PROTECTED],
Lee, John (Sydney) [EMAIL PROTECTED] wrote:
I am working on a menu to accept input from a caller like as follows:
Exten = 100,1,Answer()
Exten = 100,n,Playback(LONG-MESSAGE)
Exten = 100,n,Read(OPTION,,2)
...
When I tested it, I noticed if I start pressing a
- Original Message -
From: Lee, John (Sydney) [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 19, 2008 11:48 PM
Subject: [asterisk-users] Newbie IVR: How to read() before playback()
isfinished?
I am working on a menu to accept input from a caller like
Hello,
Do your verify, the codecs, of both clients, in your sip.conf?
What codec do you use?
Best Regards
On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay [EMAIL PROTECTED] wrote:
Hi,
I am sorry my questinos are too fundamental. I am new to Asterisk, and
hope to catch up as fast as I can.
Pete,
I have never done it but it would seem that running a SIP client on
your SIP server may be problematic.
Also, 58.251.75.228 is certainly not on your 192.168.x.x subnet. Is
your machine dual homed?
Thanks,
Steve Totaro
On Thu, Mar 20, 2008 at 9:34 PM, Carlos Rojas [EMAIL PROTECTED]
Sorry, I am tired and missed the virtual IP part. I am not quite sure
what that means or why you are sending traffic to the routeable IP.
Are you using a FQDN with external DNS or the IP in your client?
Thanks,
Steve Totaro
On Thu, Mar 20, 2008 at 10:42 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
I was trying to find out how I could put in a greeting when a caller
***first*** joins the queue.
I searched high and low but could only find (in queues.conf):
. announce, which is announcement to the agent
. announce-frequency which is announcement of queue position
.
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee, John
(Sydney)
Sent: March 19, 2008 2:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Newbie Queue: greetings when first joining queue
I was trying to find out how I could put in a greeting
I would think you'll need to do a Playback() of this message before
the
caller enters the queue, as I'm not aware of such an option provided
by
app_queue.
Exten=100,1,Answer()
Exten=100,n,Playback(greetings-earthling)
Exten=100,n,Queue(xyzqueue)
Exten=100,n,Hangup
Thanks Mark for your
(Sydney)
Sent: March 19, 2008 2:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie Queue: greetings when first
joiningqueue
I would think you'll need to do a Playback() of this message before
the
caller enters the queue, as I'm not aware
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 =
I am planning to roll out Asterisk to some offices and I have been
thinking about how to disaster proof the box.
For the production box, of course, it will have RAID 1 disk drives and
2GB memory at least.
a) If the office burns down, there is nothing much I could do.
b) If it is software error,
I am working on a menu to accept input from a caller like as follows:
Exten = 100,1,Answer()
Exten = 100,n,Playback(LONG-MESSAGE)
Exten = 100,n,Read(OPTION,,2)
...
When I tested it, I noticed if I start pressing a key before the
Playback() is finished, the input is not buffered (simply ignored)
Use Background instead of Playback, and put an
exten = XX,n,Goto
at the bottom of the context.
That should get you started.
PaulH
On Thu, 2008-03-20 at 14:48 +1100, Lee, John (Sydney) wrote:
I am working on a menu to accept input from a caller like as follows:
Exten =
Hi,
I am sorry my questinos are too fundamental. I am new to Asterisk, and hope
to catch up as fast as I can.
Problem 1:
I have my SIP client ( in one PC .102) and SIP server ( in another PC .101)
within the same land. They can make SIP connection, but when the SIP client
makes call to play
From a lot of experience - you are not being anywhere near paranoid
enough !!
Think dual RAID controllers, Dual power supplies off of, at a Minimum,
separate isolated circuits, with Hefty UPS that is in-line so it filters
the power, think mirrored systems with a simple from switch
I am trying to build a simple queue for the receptionist phone.
In other words, there is only 1 agent and that is the receptionist
phone.
I just defined a few lines in queues.conf
[console]
strategy = ringall
member = SIP/4000 ;4000 is the console extension
In extensions.conf, it is:
Lee, John (Sydney) wrote:
However, when I call from an outside line to another extension which I
then forward to 4000, I cannot get into the queue.
exten = 98786983,1,Answer()
exten = 98786983,n,Dial(SIP/4000,20)
My guess would be that extension 4000 matches somewhere else within your
On Tue, Mar 18, 2008 at 06:20:02PM +1100, Lee, John (Sydney) wrote:
I am trying to build a simple queue for the receptionist phone.
In other words, there is only 1 agent and that is the receptionist
phone.
However, when I call from an outside line to another extension which I
then forward
Robert Lister [EMAIL PROTECTED] writes:
So you either need to go a Goto(context,4000,1) or to drop it to the queue
with Queue(console) etc.
There's also Dial(Local/[EMAIL PROTECTED]). Goto is almost always a better
idea though.
/Benny
___
--
On Tue, 2008-03-18 at 18:20 +1100, Lee, John (Sydney) wrote:
I am trying to build a simple queue for the receptionist phone.
In other words, there is only 1 agent and that is the receptionist
phone.
I just defined a few lines in queues.conf
[console]
strategy = ringall
member = SIP/4000
So you either need to go a Goto(context,4000,1) or to drop it to the
queue
with Queue(console) etc.
I have chosen to use Goto(context,4000,1) from a programmer's
perspective although queue(console) works just as good.
Thanks guys.
___
-- Bandwidth and
I am using Polycom IP600 phone. If I call a phone which has DND (do not
disturb) enabled, the message to the caller will be The person on
extension ... is on the phone, please leave a message
Is there a way to pick the person ... not available message instead?
Hmmm - the issue is more that the Polycom is generating a 'busy' when it
is dialled on DND. This is not really an Asterisk issue.
You could have a look at the dial setup you are using (such as
macro-stdexten) and set it to play the unavailable message for for busy.
PaulH
On Mon, 2008-03-17 at
I am writing an extension to accept speed dial nos.
However, I forgot that these speed dials are the same for all offices
and thus would ideally be shared by offices which will host their own
Asterisk box.
I read from a few postings that this database cannot be replicated to
other Asterisk box.
I
Lee, John (Sydney) wrote:
I was thinking that if I could just do a simple copy/paste of the speed
dial records from the main database to others would be good.
You'd be better off going with a SQL setup. I have ours setup with 1
master and 2 slaves. The slaves are the remote facilities.
You could place all the information in a MySQL database, and either
reference that or replicate from that.
PaulH
On Mon, 2008-03-17 at 11:46 +1100, Lee, John (Sydney) wrote:
I am writing an extension to accept speed dial nos.
However, I forgot that these speed dials are the same for all
Does 'show features' display the correct information?
PaulH
Thanks Paul
*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #
Attended Transfer
One Touch Monitor
On our system i got:
Zap/1-1 answered SIP/106-091a2750
-- User hit '*1' to record call. filename: wav|
auto-1205385048-106-0434225491|m
Our dialplan looks like:
_0X' = 1. Dial(zap/g1/${EXTEN}||Ww)
(from show dialplan)
PaulH
Thanks Paul.
I think the problem is *1 is
I think the problem is *1 is being ignored or cannot be transmitted
successfully to Asterisk.
Finally I resolved the problem.
For some reasons, the * and 1 must be pressed pretty quickly
together on the Polycom phone before it can be transmitted successfully
to Asterisk.
I think I cannot deny
My guess (from your features) is that the * for disconnect and *1 for
records are clashing - maybe set disconnect to **73 to avoid this.
And - yes, it can be tuned:
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation (default
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