I appreciate it if someone can post an an example for function CHANNELS showing
the usage of the regular expression filter.
Basically I would like to get a count of active channels having a certain
criteria. Is it possible to search for a channel having a custom variable set
to specific value
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ringing in queues
On 13 March 2015 at 14:04, Matt Hamilton efes9...@hotmail.com wrote:
We use the ringall strategy for a small queue with 4 members. When a call comes
in, if one of the members is busy, all the phones except
We use the ringall strategy for a small queue with 4 members. When a call comes
in, if one of the members is busy, all the phones except the busy phone rings
(as intended). While the other phones are ringing, if this busy phone becomes
available again, we would like to have it start ringing.
option in asterisk.conf.
On Tue, Oct 29, 2013 at 6:17 PM, Matt Hamilton mistral9...@hotmail.com
wrote:
Is there a limit to the number of parked calls Asterisk can handle?
Thanks,
Matt
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Is there a limit to the number of parked calls Asterisk can handle?
Thanks,
Matt
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New to Asterisk? Join us for a
We are planning to have about 100+ parking lots defined in features.conf , each
with about 4 unique park positions. Asterisk will be handling all the parking
and unparking (we don't exclusively use Park/ParkedCall in the dialplan):
[parkinglot_a]
parkpos = 1-4
context=parked
[parkinglot_b]
Is there a way to find out the no of parked calls in a parking lot by name (in
a multiple parking lot environment) from within the dialplan (not CLI) other
than writing a custom function (like VMCOUNT)?
Thanks,
Matt
--
I'm trying to implement call parking with asterisk and Cisco SPA504G phones:
features.conf
parkext = 700
parkpos = 701-702
context = parkedcalls
I defined one of the unused keys to park the calls:
Key2:
fnc=sd;ext=700@10.0.1.103;vid=1;nme=Park
I also defined two other keys to pickup/unpark
I'll answer my own question:
Setting Keep Referee When REFER Failed to Yes on the Cisco phone seems to do
the trick.
From: mistral9...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Sun, 20 Oct 2013 11:56:17 -0400
Subject: [asterisk-users] Call parking issue with Cisco SPA phone
I'm
I'm trying to implement parking with only one button to park and unpark a call.
Scenario:
Call is answered, I press the button (on a Cisco SPA504) to park the call, it
comes to [from-office] context where the call is parked successfully (there is
no parking lot number announcement though).
Parking/unparking will be done from multiple phones so that someone else can
pickup/unpark the call from their phone that I parked on mine. I'm just testing
it on one phone now.
I'm trying to simulate the SLA functionality (which Asterisk has, but it's not
very scalable and they haven't
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Matt Hamilton
Sent: Monday, October 14, 2013 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] parking - why doesn't this work?
Parking/unparking
We have been using Asterisk SLA for a while with Cisco SPA series phones. Once
in a while the phones flash, but not ring when a call comes in. We can pick it
up and talk to the caller even if that's the case.
This is pretty random (might not happen for couple of weeks). The quick
solution is
Have you tried restarting the phone instead of Asterisk? I don't think that
Asterisk sends
separate commands to the bell and to the call LED. Since the LED is flashing,
it is likely that
the SIP INVITE signal from Asterisk is ok. Also the ring tone normally does
not come from
We have a couple of cisco SPA phones and 3CX softphones behind a NAT firewall
in a remote location. Firewall is connected to a bridged router which connects
them to the public internet.
Router 5.6.7.8
Firewall 5.6.7.9(gateway 5.6.7.8)
Cisco SPA phone 192.168.1.4
Softphone
remember correctly it's an FCC mandate that a number cannot be
ported within 30 days of a previous port.
Greg
Matt Hamilton mistral9...@hotmail.com wrote:
Don,
Inbound reliability is very important. We don't use toll-free numbers, but
we will look into that. I thought porting numbers
We have couple of stores with published phone numbers. We are currently using a
telecom company that hosts those numbers and provides us with SIP trunking.
Recently we experienced couple of outages with this company, so we are looking
into getting a backup provider just in case.
As far as I
The process will depend on your provider, of course, but I know some
have an option that if they are unable to reach
your box, then they can
auto-forward to another DID, or to a voicemail box, or to a user-defined
function, etc.
Forwarding to another DID will/should work for us assuming they
switched between the sip providers.--Don From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton
Sent: Tuesday, April 30, 2013 10:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
Date: Thu, 27 Sep 2012 10:23:35 +0200
From: lenz.lo...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR
I'd go for MyISAM and would set up a remote replica if data integrity is
important.
If you have like 1000 calls of (say) 30 seconds
Which one (InnoDB or MyISAM) is preferred for CDR as far as write performance
is concerned?
Thanks,
Matt
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New to
Our top priority is the raw Write (INSERT) performance, Read (SELECT)
performance is not important. Strict ACID compliance is not necessary either.
MySQL (on a separate database server) should be able to handle inserting CDR
records (approximately up to 10 records for each call) for about 1000
I would like to have the SLA hints created on the fly as the phones register
dynamically. Is this possible?
The static version that works:
[sla_stations]
exten = 10041_ln10041,hint,SLA:10041_ln010041
exten = 10041_ln10042,hint,SLA:10041_ln010042
exten =
ConfBridge is the preferred conference application in Asterisk 10+. While
MeetMe is currently deprecated, you can still enable it and run it in
Asterisk 10+.
What's going to happen to SLA (which is heavily integrated with MeetMe)? Will
the functionality be ported to ConfBridge?
Thanks,
Thanks Paul, we are looking into the testsuite.
Date: Sun, 3 Jun 2012 18:32:57 -0400
From: paul.belan...@polybeacon.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Automate SLA testing
On 12-06-03 03:56 PM, Matt Hamilton wrote:
We would like to automate Shared
We would like to automate Shared Line Appearance testing (e.g. phoneA answers a
call, puts in on hold, phoneB picks up the call on hold) in our lab. Are there
any tools/SIP call generators/clients that may help us create such a scenario?
Thanks,
Matt
Thanks Ishfaq, I need something from within the dialplan though.
From: i...@pack-net.co.uk
To: asterisk-users@lists.digium.com
Date: Wed, 18 Apr 2012 10:06:38 +0100
Subject: Re: [asterisk-users] device state of a realtime queue member
On Tue, 2012-04-17 at 11:53 -0400, Matt Hamilton
-0500
Subject: Re: [asterisk-users] device state of a realtime queue member
You can use system() to do this from the dialplan From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton
Sent: Wednesday, April 18, 2012 11:42 AM
I'm trying to find if a realtime queue member is paused or not from the
dialplan.
For a paused, not in use phone, DEVICE_STATE returns not in use only. Is
there a function that will tell if the phone is paused or not (other than
querying the database directly)?
Thanks,
Matt
Is it possible to have a meetme conference timeout (and go to the next line in
the dialplan) if there is only one participant left?
Thanks,
Matt
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10:22:49 -0500
From: kpflem...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SendText causes Retransmission errors
On 03/16/2012 09:43 AM, Matt Hamilton wrote:
Hi,
I'm using SendText to send a text message when the user picks up a line
in a SLA setup (even
Hi,
I'm using SendText to send a text message when the user picks up a line in a
SLA setup (even though I'm not sure the problem is related to SLA). I'm on
Asterisk 10.2.1 (same in 1.8.9)
[from-office]
..
same = n,SendText(hi)
same = n,SLAStation(line1234)
..
Here is a simplified version of
Hi,
Is it possible to transfer a call in Asterisk SLA (shared line appearance)
e.g., call comes in via SLATrunk, it's answered, and transferred to an outside
number?
Thanks,
Matt
--
_
Hi,
For REGISTER and SUBSCRIBE requests coming from UACs, is it possible to make
Asterisk skip authentication even if a secret is defined in sip.conf for the
peer; i.e. similar to insecure=invite for INVITE requests?
If I leave secret blank, Asterisk doesn't require any authentication - this
Date: Tue, 14 Feb 2012 09:44:38 -0600
From: kpflem...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] skip authentication for REGISTER
On 02/14/2012 08:43 AM, Matt Hamilton wrote:
Hi,
For REGISTER and SUBSCRIBE requests coming from UACs, is it possible
Hi,
Are there any guidelines/recommended values for setting the registration
expiration and subscription expiration for SIP phones?
The default values for those settings on our phones are 60 secs. Any
disadvantages for making them longer; e.g. 300 secs or more?
Thanks,
Matt
2012-01-20 20:09, Matt Hamilton skrev:
Hi,
Once in a while when a SIP channel connected to meetme
conference is hung up, I start getting the following error
multiple times:
WARNING[14031]: app_meetme.c:3668 conf_run
Hi,
Once in a while when a SIP channel connected to meetme conference is hung up, I
start getting the following error multiple times:
WARNING[14031]: app_meetme.c:3668 conf_run: Unable to write frame to channel
Local/100203@h
The status of the channel is not updated, and the only way to get
Hi,
I have an Opensips server dispatching to 3 Asterisk servers. I would
like to assign public IPs to all of these servers and avoid NAT
altogether - phones will also have public IPs. The way I set this in the
lab, all the SIP traffic goes thru the SIP proxy (Opensips) and RTP
goes
I have a very basic setup where a UAC registers with Asterisk 1.8.7.2 - both on
the same subnet, no nat.
The following is the flow of messages:
1. UAC sends the registration request
2. Asterisk responds with 401 Unauthorized with a new nonce
3. UAC sends a new digest with the nonce received
Asterisk destroys SIP dialogs in 32 secs, so increased the UAC registry
expiration to 300 secs just in case, but that didn't help either.
From: mistral9...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Sun, 1 Jan 2012 18:13:07 -0500
Subject: [asterisk-users] registration not authorized
I have a couple of performance/memory related questions:
Is there any downside to using long URIs as far as memory or database (mysql)
performance is concerned, e.g.
sip:1234567890_1234567...@abc.com? Or is this negligible?
Also is there a performance hit if no pattern matching is used?
Thanks Sebastian. It was a phone related issue. Factory resetting the phones
and reconfiguring them fixed it. It probably was a CW issue as you suggested.
I think it is up to your phones to allow only one concurrent session,
you could check call-waiting is deactivated on your phones?!
If
I have a queue that distributes calls among 3 phones. When a phone is in use
(including on hold), queue skips that device and sends the call to the next
available one as expected. On the other hand, if a call comes in while one of
the phones is ringing, the queue doesn't seem to recognize that
Is it possible to load some parts of the extensions.conf file via static
realtime?
For example,
extensions.conf
[some_context]
#include abc.conf
extconfig.conf
abc.conf = mysql,asterisk,ast_config
So far, I wasn't able to get it going - Asterisk crashes at startup. Maybe
abc.conf
I integrated Opensips with Asterisk Realtime (Asterisk sipusers/peers
point to Opensips subscribe table via a view). Opensips handles the
registrations. However, when a call comes in
(INVITE is routed to Asterisk), it seems like Asterisk doesn't know
about the user (or sees the users as not
Thanks Dale, you pointed me in the right direction.
You are calling the Dial() application here. If you are using
queues, you should use the Queue() application.
I'm using Local channels with the queue:
- queues.conf
[support]
member = Local/1001@handle-queue
Thanks Danny.
[clearkeys] Exten = start,1,answer() Exten = start,n,dbdeltree(foo)
Exten = start,n,hangup Set and retrieve Global variables for small
searches.
I will try the local call option to [clearkeys].
I guess I can also use a global flag to call dbdeltree only once in the
Mike,
Just enter a unique systemname into asterisk.conf for each box. This system
identifier is appended to the front of the unique id field in cdr.
/etc/asterisk/asterisk.conf
[options]
systemname=asterisk1
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 21 Nov
@lists.digium.com
Subject: Re: [asterisk-users] Deleting AstDB family at start
On 11-11-21 03:46 PM, Steve Edwards wrote:
On Sun, 20 Nov 2011, Matt Hamilton wrote:
Is it possible to delete the keys belonging to a family in AstDB at
Asterisk startup? I would like to repopulate it from
Have you tried, instead of pre-processing the caller before calling
Queue(), checking the ${QUEUESTATUS} variable.
Even when the phones are UNREACHABLE, QUEUE is still trying until it times out
- ${QUEUESTATUS} = TIMEOUT
I get the following for all the members of the queue, in a loop,
Is it possible to delete the keys belonging to a family in AstDB at Asterisk
startup? I would like to repopulate it from another source each time Asterisk
is restarted.
I know there is a DBdeltree(family) function. Is there a context that only
runs once (automatically) at Asterisk startup
I would like to perform 2 checks on a queue:
1. if the caller stays in the queue for a certain time, I would like to forward
him to phone A.
2. if the devices/members in the queue are not reachable, I would like to
forward him to a phone B.
The first one is straight-forward via the timeout.
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