On 09/14/2012 12:45 PM, RSCL Mumbai wrote:
I need a list of calls Answered and Disconnected in less than 5 sec.
http://dev.mysql.com/doc/refman/5.6/en/select.html
http://www.google.com/search?q=sql+tutorial
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On 08/28/2012 01:51 PM, Olivier wrote:
Let say I cannot touch the files in which those 2 instructions are set:
[timeconditions-toggle]
exten = *2711,hint,Custom:TC11
...
[ext-local]
exten = 6452,hint,SIP/6452
...
Then what can I do allow a given SIP phone to successfully subscribe
to both
On 08/27/2012 01:02 PM, motty.cruz wrote:
Hello, would like to have distintive ringtone for internal calls, google
gave me blurr answer.
My extensions are 46**, any calls made within 46** I want to ring
differently than external calls.
Assuming you are using SIP handsets, distinctive ring
On 08/23/2012 10:05 AM, Jonas Kellens wrote:
Hello,
using asterisk 1.8.11.1
using realtime queues
When trying to remove a queue member, I get the following :
-- Executing [122@from-TESTCORP:2]
RemoveQueueMember(SIP/testcorp5-000c,
testcorpq1,SIP/testcorp7) in new stack
WARNING[18788]:
On 08/22/2012 03:41 PM, Giuseppe Longo wrote:
Is it better Asterisk 11, right?
At least 9.2 better, for sure.
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On 08/21/2012 02:52 PM, Noah Engelberth wrote:
The short of the output is -- there is no console output showing ==
Extension Changed 302[hints] new state on the Ringing or InUseRinging
events -- only on InUse or Idle events (which matches what I'm seeing
on the phones).
Weird. I just did a
On 08/20/2012 03:20 PM, Noah Engelberth wrote:
And after 303 tries to call 302 while 301 302 are still on a call
(301 302 on a call, plus 303 calling 302):
-= Registered Asterisk Dial Plan Hints =-
_3XX@hints : Custom:${EXTEN} State:Idle
Watchers 0
-0013)
SIP tracing shows the response from the phone as:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 172.20.20.6:5060;branch=z9hG4bK6ad7c9fd;rport=5060
From: Phil Frost sip:207@172.20.20.6;tag=as719c88e2
To: sip:pfrost@172.20.25.126:3072;line=l1no5zvm;tag=y5f8ddjzb0
Call-ID
On 08/17/2012 10:09 AM, Phil Frost wrote:
On 08/17/2012 02:28 AM, Olle E. Johansson wrote:
If a call is forwarded and hit the dialplan again, it's forwarded to
the context set in the channel variable FORWARD_CONTEXT.
So you could set this variable before you hit queue(), then do things
I'd like to allow my users to forward their calls using the forwarding
feature on their SIP handsets and continue to receive Queue() calls.
Currently I set the 'i' option in Queue() so that if a user forwards to
their cell phone, or any other extension that has voicemail, the
voicemail doesn't
On 04/04/2012 04:50 PM, Olivier CALVANO wrote:
Hi
i am search a solution for change the number called.
Sample:
I have a Linksys SPA942 connected in SIP with my server.
When this phone call a number: 043112
automatiquely change in 3343112
because my carrier want a number in
On 04/02/2012 08:35 PM, Warren Selby wrote:
On Mon, Apr 2, 2012 at 7:05 PM, Paolo Supino paolo.sup...@gmail.com
mailto:paolo.sup...@gmail.com wrote:
Hi
A couple of weeks ago I asekd how to setup a fallback numer and one of
the reply I received was to se GotoIF and ${DIALSTATUS}.
means it can't do things like intercept and act on DTMF or monitor the
call.
Native bridging is when media is forwarded with Asterisk, but for whatever
reason (different codecs, maybe) Asterisk must inspect or modify the stream.
Could mean a significant CPU load.
--
Phil Frost
Macprofessionals
) ]
conn=0x2089780, query='select 1'
[ fetched 1 rows ]
Not really sure what attribute 113 is or why it couldn't be set, but since I
see a successful select 1 at the end, I'm thinking it's benign. Am I missing
something obvious? Any ideas on where to look next?
--
Phil Frost
Macprofessionals
On Mar 21, 2012, at 08:36 , Andrew Latham wrote:
On Wed, Mar 21, 2012 at 8:27 AM, Paolo Supino paolo.sup...@gmail.com wrote:
Hi
I was asked by our development departement to setup asterisk in a
manner that if someone calls an extension in the department that was
was only configured, but a
All my phones have call waiting, so it's unlikely DIALSTATUS ever gets set to
BUSY. So, I'm trying to decide what to do about the two greetings users record,
busy and unavailable.
If I could, I could just disable one. Then there's only one greeting, and no
chance for confusion. I could
On Feb 23, 2012, at 18:44 , Bryant Zimmerman wrote:
I was working on this today. I have it figured out but I don't have simple
dialplan code I can share as we are doing a lot of external db and script
calls to make ours work with our realtime stuff. We also are using the
Dynamic Parking
I'm trying to emulate the functionality of our existing phone system, which is
somewhat different than what Asterisk provides with a trivial parking
configuration. I'd like each user to have three park buttons, park 1, park 2,
park 3. The snom 870s I'm using have a Park+Orbit button, which best
On Feb 23, 2012, at 16:32 , Richard Mudgett wrote:
exten = _*70[123],1,NoOp(parking in ${EXTEN:1})
same = n,Set(PARKINGEXTEN=${EXTEN:1})
same =
n,GotoIf(${DEVICE_STATE(park:${PARKINGEXTEN}@parkedcalls)}=INUSE?busy)
same = n,Park()
same = n(busy),Busy()
What I'm hoping to
On 02/22/2012 07:26 AM, virendra bhati wrote:
Does anyone know the correct information of my question. All are
move round and round .
Why don't you make a call, and watch the network traffic with tcpdump,
wireshark or similar? Wireshark in particular has an analysis module
that will show you
On 02/22/2012 08:01 AM, virendra bhati wrote:
*Will these port of UDP, RPT [assume you mean RTP] or Both ?*
It's evident from your response that you do not have a solid
understanding of networking fundamentals. The full answer to your
question will quickly go out of scope of this list and
On Feb 21, 2012, at 17:01 , Phil Frost wrote:
On Feb 14, 2012, at 17:13 , isr...@gmail.com wrote:
On the snom too
Create a conferance and then press the transfer button. That will join the
parties and release the receptionist
Hmm...You can do that with just hitting the transfer button
On Feb 14, 2012, at 17:13 , isr...@gmail.com wrote:
On the snom too
Create a conferance and then press the transfer button. That will join the
parties and release the receptionist
Hmm...You can do that with just hitting the transfer button, or is there more?
I'm using a Snom 870 with
I'm attempting to pull SIP users from LDAP, following the instructions from
here:
http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
However, when I attempt to register a user from LDAP, I see on the console:
chan_sip.c:24431 handle_request_register:
Asterisk and Avaya, but not from Asterisk, through Avaya, to
our SIP trunk. Also, I haven't yet gotten the caller ID to be what I want.
I can share more detail of my configs, but first, you didn't say what kind of
Avaya you have. IP Office? Or something else?
--
Phil Frost
Macprofessionals
I'm wondering how one might implement a transfer where a receptionist
introduces a caller to the recipient in a 3-way conference before hanging up,
leaving the other two parties connected. Something like this, from the
perspective of the customer:
Customer: Hi. I'd like to buy a widget.
On Feb 14, 2012, at 15:34 , Danny Nicholas wrote:
As I read this, this is a regular attended transfer.
No, as I understand an attended transfer, there is no 3-way period where the
receptionist introduces the caller to someone else. In an attended transfer,
from the caller's perspective, he's
I'm trying to implement a very simple call queue for a small, low volume
helpdesk. We have 2-5 agents, and rarely does the queue get more than 1 or 2
callers deep. I'm using the ringall strategy and I want calls answered in FIFO
order.
Say caller A calls the queue, and there is one member
On Feb 10, 2012, at 14:37 , Phil Frost wrote:
I'm trying to implement a very simple call queue for a small, low volume
helpdesk. We have 2-5 agents, and rarely does the queue get more than 1 or 2
callers deep. I'm using the ringall strategy and I want calls answered in
FIFO order.
Say
I'm attempting to configure an H.323 trunk (using chan_h323) between an
Asterisk box and an Avaya IP office. It mostly works. Calls from Polycom SIP
devices registered to Asterisk can place calls over the trunk to IP Office
extensions and everything works great. However, calling from an IP
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