From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UxBoD
Sent: Tuesday, 12 January 2010 17:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multi-Tenant Parking
Should that not say parkinglot
Have you looked at this?
http://www.google.com/#q=app_valetparking
I have - but would rather use the inbuilt functionality if possible before
resorting to third-party code...
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Computer viruses - It is your responsibility to scan this email and any
I have found that this seems to be a functional difference between the Park()
and the ParkAndAnnounce() functions. Park() respects the parking lot
specification, yet ParkAndAnnounce() does not respect the fact that you’ve
tried to arbitrarily set the parking lot. The code below “works” as
thinning on top - don't want to lose any
more hair on this one!
Michael Wyres
Technical Specialist
Communications Design Management
Level 1 / 99 King St
Melbourne Victoria 3000
P + 61 3 9601 6600
F + 61 3 9601 6601
mwy...@cdm.com.aublocked::mailto:sbro...@cdm.com.au
[cid:image001.jpg
Would it not be easier for you to just bill them for access to 12 channels (6
extensions x 2 channels each)? Seems simpler. Then bill them for the calls
they actually make.
Then set call-limit=2 for each extension in sip.conf?
See:
Is it a single user? Or every single phone?
If it's a single user, and you can get hold of a UPS with power conditioning on
it, try plugging the various devices into it - there might be some dirty power
coming along.
From: asterisk-users-boun...@lists.digium.com
I would without the deny and permit directives in the SIP, and rule out
some sort of clash there that is rejecting the address the registration is
coming from, and take it from there.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
%20 usually represents a space in escaped URL format - perhaps you've
inadvertently got a space in front of the username in the SIP account on the
e71?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean
To be perfectly complete, exactly which inbound ports to open will depend on
the phones in use. For example, a Cisco 7940 (using this example because I
have one on my desk at the moment), the default ports from the config are:
voip_control_port : 5060
start_media_port : 16384
end_media_port :
Hi Travis,
There's lots of different ways to attack on-call roster solutions in Asterisk
- as Danny suggested, FollowMe() is definitely an option (and normally the
best), but it doesn't always suit the business need. However, also as Danny
suggested, in most cases using ASTDB in some way to
trying to achieve.
Thanks Danny Michael,
Travis
- Original Message -
From: Michael Wyres mwy...@cdm.com.au
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, November 16, 2009 2:23:49 PM
Subject: Re: [asterisk-users] Queues
Hi Travis
Sometimes they reboot when you try this, but usually not - but you can just
change one setting in the network configuration (eg: change the phones IP
address), and it will go through just the very last part of it's normal boot
process, and re-pull it's TFTP configuration, and update things -
Throwing him off the list would not achieve anything - he still has our email
addresses, and will still be able to send you email.
Unless of course, you pop his email address on the DENY list of your
gateway...*whistles innocently*
From: asterisk-users-boun...@lists.digium.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard
Sent: Friday, 13 November 2009 06:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] allowguest defaults to
The reasons for poor call quality are many and varied.
As another poster suggested, the headset you are using might be poorly
configured, or just a poor example.
An under-spec server could also do it - I use two simple, low-spec Virtual
Machines in my dev lab that I bring up when I want to
Have you tried nat=yes in the definition in sip.conf?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Thursday, 12 November 2009 13:30
To: asterisk-users@lists.digium.com
Subject:
Try:
[tutorial]
exten = 1234,1,Dial(SIP/gianca,10,t)
exten = 12345,1,Dial(SIP/giusy,10,t)
You want a / between SIP and the name of the phone, not an ,.
The 10 refers to the number of seconds you want the phone to ring. The t
allows the channel to be transferred after pickup - not strictly
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