Rob Schall wrote:
Perfect. Here's a quick and hopefully doable followup question. We have
Polycom Soundpoint 501 phones. Is there a way to have a phone check 2
voicemail boxes? If we have a queue, and we want the MWI to show for say
that users's extension 1000 and the special billing vm box of
yusuf wrote:
j wrote:
Greetings!
I've searched far and wide for an answer and have gotten no where, so I
was hoping one of you guys might have the answer;
Is it possible to dynamically add a context to the dialplan?
You can add extensions via the CLI, however if the context doesn't exist
I
Time Bandit wrote:
Significant albeit insanely stupid Asstricks message:
2007-01-30 09:22:57 DEBUG[9946]: pbx.c:2300 __ast_pbx_run: Oooh, got
something to jump out with ('2')!
(Oooh how about creating errors we can figure out Digium!)
Any thoughts
What Error ? it says DEBUG
This just tell
Rob Schall wrote:
Hello all,
Lee recommended QUEUESTATUS, but that seems to return if anyone is in a
specific queue, and not if the current call came from a queue. I
probably just misunderstand how it all works. :)
Thanks all!
Rob
Hi Rob,
Remember that I am pretty new to Asterisk myself.
Lee Jenkins wrote:
Rob Schall wrote:
Hello all,
Lee recommended QUEUESTATUS, but that seems to return if anyone is in a
specific queue, and not if the current call came from a queue. I
probably just misunderstand how it all works. :)
Thanks all!
Rob
Hi Rob,
Remember that I am pretty new
Shane Spencer wrote:
Reload.. Reload.. Reload..!
LOL.
--
Warm Regards,
Lee
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Rob Schall wrote:
A question about Queues and Dial Plans
We are trying to set up a customer service queue. I've set up the queue
and agents who will participate. However, there's still one area I'm not
sure how to make it work. After 60 seconds, I need it to decide that no
one is available,
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I do get
congestion, but that is expected. However, when I try to
Carlos Rojas wrote:
Hello,
Do you include in your zapata.conf
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
There are any problems with hang up
I tried adding these parameters as you suggested, but then was unable to
dial out at all. Removing them allows me to dial out again, but
Lee Jenkins wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I do get
congestion, but that is expected
Lee Jenkins wrote:
Lee Jenkins wrote:
After playing around a bit, it appears that this is just random as far
as I can see. It may allow me to dial a few times, but then hangup.
After rebooting my server, it may let me dial once and then start
hanging up.
I really hope it's not a fight
Eric ManxPower Wieling wrote:
Lee Jenkins wrote:
I forgot to mention that the one thing that seems to be consistent is
that I can get the zap line to reset and dialout again correctly by
calling into the system on that zap line, dialing and extension and
allowing the extension to hangup
Stephen Bosch wrote:
Hi, Lee:
Lee Jenkins wrote:
Lee Jenkins wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I
Stephen Bosch wrote:
Lee Jenkins wrote:
I forgot to mention that the one thing that seems to be consistent is
that I can get the zap line to reset and dialout again correctly by
calling into the system on that zap line, dialing and extension and
allowing the extension to hangup on the caller
Stephen Bosch wrote:
Hi, Lee:
Lee Jenkins wrote:
Hi Eric,
I do not have any extensions with wildcard patterns like that. I am
trying my local 7 digit cell phone (tried other patterns though and same
result). Example:
exten=_9NXX,1,Macro(DialOutside,ZAP/3/${EXTEN:1})
I thought
Eric ManxPower Wieling wrote:
Lee Jenkins wrote:
Stephen Bosch wrote:
Lee Jenkins wrote:
I forgot to mention that the one thing that seems to be consistent is
that I can get the zap line to reset and dialout again correctly by
calling into the system on that zap line, dialing and extension
Lee Jenkins wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I do get
congestion, but that is expected
OK, I think I may have found the problem for myself at least. Actually,
a friend of mine suggested it. Apparently, Asterisk is a little too
fast for the card.
Placing a w in front of the number to insert a pause looks like it did
the trick!
Dial(ZAP/1/w555)
Looks like it gives the
Lee Jenkins wrote:
OK, I think I may have found the problem for myself at least. Actually,
a friend of mine suggested it. Apparently, Asterisk is a little too
fast for the card.
Placing a w in front of the number to insert a pause looks like it did
the trick!
Dial(ZAP/1/w555
Stephen Bosch wrote:
Hi, Lee:
Lee Jenkins wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I place an outbound call on the Zap line (Zap/3), everything works
fine except when the called party hangups before I do. I do get
congestion
Yuan LIU wrote:
From: /Lee Jenkins [EMAIL PROTECTED]/
[...]
If I call out to a party on that Zap line and hangup first, I do
not experience that problem. It looks like Asterisk is not getting
the termination signal from the telco (Verizon) when the other
party hangs up first.
Running
Rizwan Hisham wrote:
Whats the difference between the following statements in extensions.conf
include=inbound
AND
#include inbound/*.conf
Hi, checkout this page:
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf
With the #include filename statement in extensions.conf,
Matthew Rubenstein wrote:
The H.264 codec patent by Qualcomm has been ruled invalid by a San
Diego Federal jury:
http://www.eetimes.com/news/semi/showArticle.jhtml?articleID=197001066 .
That means that H.264 codecs can now be written, distributed and revised
freely under any license
Lee Jenkins wrote:
Hi all,
I've just setup a sip line with Telasip and when they route the calls to
my asterisk box, they include an extension along with the context that
is defined in sip.conf for that DID.
At first, I couldn't figure why they were getting 404 error from my
asterisk box
Michiel van Baak wrote:
On 17:36, Thu 18 Jan 07, Lee Jenkins wrote:
Michiel van Baak wrote:
exten=999,1,Queue(support,tr|||60)
and never put it back. From there it was a downward spiral ;)
If you remove the r, does that fix the issue ?
Yes. It did. Still not sure why it didn't work
Lee Jenkins wrote:
Hi all,
Where can I find the status definitions for the status field below?
Googled /web/voip-info.org and can't seem to find anything.
Event: QueueMemberStatus
Privilege: agent,all
Queue: support
Location: SIP/111
Membership: dynamic
Penalty: 0
CallsTaken: 0
LastCall: 0
Hi all,
Where can I find the status definitions for the status field below?
Googled /web/voip-info.org and can't seem to find anything.
Event: QueueMemberStatus
Privilege: agent,all
Queue: support
Location: SIP/111
Membership: dynamic
Penalty: 0
CallsTaken: 0
LastCall: 0
Status: 1
Paused: 0
Hi all,
I'm implementing call files and everything works nicely except that the
variable that I set in the call file does not seem to get populated.
Channel:SIP/MyProvider/910555
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context:myCallFileContext
Extension:
Lee Jenkins wrote:
Hi all,
I'm implementing call files and everything works nicely except that the
variable that I set in the call file does not seem to get populated.
Channel:SIP/MyProvider/910555
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: myCallFileContext
Hi all,
I've just setup a sip line with Telasip and when they route the calls to
my asterisk box, they include an extension along with the context that
is defined in sip.conf for that DID.
At first, I couldn't figure why they were getting 404 error from my
asterisk box, but then figured
Hi all,
I have configured the queue below, but when I go into the queue,
asterisk does not announce hold time:
[support]
musiconhold=default
strategy=ringall
context=check_time
timeout=20
wrapuptime=1
maxlen=3
announce-frequency=5
announce-holdtime=yes
joinempty=no
leavewhenempty=yes
Bernardo Vieira wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I think what you're looking for are configuration templates:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+template
Then you would have something like:
[grandstream]
context=default
type=friend
qualify=yes
Michiel van Baak wrote:
What's the line in extensions.conf to go into the queue ?
I found out that if you use the r flag there (provide
ringtone) the announcements wont work.
Odd. I *did* originally have it set to use MOH:
exten=999,1,Queue(support,t|||60)
but it didn't work (for whatever
Yehavi Bourvine +972-8-9489444 wrote:
Enclosed bellow is the fragment from extenstions.conf which does two things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi:
That's a very nice little script.
--
Warm Regards,
Richard Soderblom wrote:
Hi.
Has anyone had any success in implementing a callback or ringback
function in Asterisk?
I've had a look at the callback-voicemail example on voip-info.org
http://www.voip-info.org/wiki/view/Asterisk+tips+callback
However it won't quite work for me.
I need it for
[EMAIL PROTECTED] wrote:
Hello all,
we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent.
For example there are 10 callers in the queue, an Agent
Kate Kretz wrote:
Dear Sirs,
I'm looking for a tool which can do the following:
1) higher level of administration, only one person, it can create
domains and per-domain administration accounts
2) lower level of administration, many persons, each can add new
extensions and change passwords
Kate Kretz wrote:
actually, I was looking for Web thing. I'd like to delegate my customers
(i.e. companies) to manage their extensions via Web.
what are those *.ael files ?
let me desribe the task more precisely. we run telecom, and we sell
phone numbers to companies. what do we want to do
Eric ManxPower Wieling wrote:
Steve Edwards wrote:
On Thu, 11 Jan 2007, Yuan LIU wrote:
AGI doesn't see the name var; all it sees is an array @ARGV (or
whatever in the respective language). As the documentation says,
values are passed like command line arguments.
But, in the interest of
Ralph Liebessohn wrote:
Hi,
I'm trying to write a AGI in PHP to get the numbers dialed (with
read()), save it into a variable to insert it into a SQL server
database. But I cannot see results into the variable, it always return
NULL.
Here is a piece of the AGI.
fwrite(STDOUT,exec Read
Ralph Liebessohn wrote:
Hi Lee,
thanks for the tip. I tried other methods trying to get the variable
value, but no success.
Doing a GET VARIABLE my_var after READ the get variable returns the
value I dialed, but doesn't give the exact value to it. I got Resource
ID #1 instead.
Using:
Ralph Liebessohn wrote:
Using:
fwrite(STDOUT,exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
\n);
fwrite(STDOUT,get variable my_var \n);
fflush(STDOUT);
$my_var=STDIN;
fwrite(STDOUT,exec saydigits $my_var \n);
I got it:
Also you might try concatenating
Olivier wrote:
By Trixbox, do you mean FreePBX (formely AMP) ?
Yes. Sorry, I tend to group them together and should have noted that
they are separate products.
--
Warm Regards,
Lee
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Hi all,
I just finished writing the bulk of an AGI interface to FirebirdSQL
databases and I noticed that when assigning a variable through AGI (I
assume this also applies within the dialplan), you have to enclose it in
quotes if there are any space.
Does Asterisk strip off the quotes when
Trevor Peirce wrote:
Lee Jenkins wrote:
Does Asterisk strip off the quotes when storing the value?
You could do a 5 minute test to figure that out...
I am not at my house tonight where I have access to a box and was
wondering about it on the car ride over to where I am.
Of course, you
Olivier wrote:
Hi,
For a 20 users prospective customer, I'm wondering if any GUI would
allow and end user to edit an Asterisk IVR tree ?
For instance, I'm looking for something allowing to edit interactions like :
wait up to 20 seconds and say this to reach sales department, type 1,
to
Bruce Reeves wrote:
After skimming over your readme file I thought I would ask, how does
this app differ from passing the parameters to the swift program using a
System dial plan command? You mention having cepstral play back a text
file in a certain voice, which I have done from the dialplan
Thanks to all for the feedback. I have created a wiki page here:
http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper
http://preview.tinyurl.com/yl9utq
I will host it on my company website for now. Seems like a small
project to bother with SF.net.
--
Warm Regards,
Lee
Josué Conti wrote:
Always...
Desire that in the New Year that if you really initiate...
It hears the words that always it desired to hear. It pronounces the
phrases that one day it desired to repeat.
It feels the emotion that always waited to feel.
It walks for the tracks that one day it
Lee Jenkins wrote:
After figuring out a problem with AGI and freepascal, I have finished
writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a
small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt.
I'd like to give it to the community (source/binary
Hey all,
After figuring out a problem with AGI and freepascal, I have finished
writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a
small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt.
I'd like to give it to the community (source/binary) and was
Leo Ann Boon wrote:
Have you tried using the agi unit at
http://home.cogeco.ca/~camstuff/agiunitpas.txt?
Leo
Yes. I have tried that and get the same thing.
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Warm Regards,
Lee
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Lee Jenkins wrote:
Hi all, after trying a number of different ways to get this to work, I
have found a way.
For some reason, it does not appear to me that using standard Writeln()
to send commands to Asterisk are ignored for some reason, even when
appending #10 or #13#10 or #13
Matthew Mackes wrote:
Check them out:
www.neobits.c_m/zultys_wip_2__wi-fi_ip_voip_sip_telephone_p9656.html
Got 404 error on that link.
--
Warm Regards,
Lee
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Hi Everyone,
I'm wondering if anyone here write AGI's in compiled binaries. I'm
writing a small Cepstral AGI in Freepascal/Lazarus. I know there are
some other AGI's out there, but I wanted to add some more functionality
than what is available such as having the AGI determine if the data
Moises Silva wrote:
use agi debug command from the Asterisk CLI to see what is going on.
Also, the last time I checked, \n is needed at the end of any
command sent to Asterisk.
Regards.
Hi, sorry I have already done that, but did not mention it. The output
that is displayed when I turn
Hi all,
I am getting the following popping up in my asterisk CLI. Everything
seems to working ok, but I'm curious as to what exactly these messages mean:
Dec 23 12:28:44 NOTICE[9858]: chan_sip.c:11123 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from
Doug Lytle wrote:
Lee Jenkins wrote:
Dec 23 12:28:44 NOTICE[9858]: chan_sip.c:11123
handle_request_subscribe: Got SUBSCRIBE for extension
[EMAIL PROTECTED] from 192.168.1.104, but there is no hint
for that
If I'm remembering correctly, it's a message you'd get if you had a
Polycom
I was playing with aelparse last night and I thought it would be nice if
the output of the it's operation was a little more structured.
I've written a app that allows me to edit ael/conf files from a windows
environment and upload them to the asterisk box, commit a reload,
restart, etc,
Tzafrir Cohen wrote:
Maybe an optional different file descriptor rather than a dump file?
Would that have been of more use to you?
That could certainly work.
--
Warm Regards,
Lee
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Eric Jacksch wrote:
You might also want to look at what the legal situation is in your
jurisdiction. Here one only needs the consent of one party to the call,
so I don’t have to advise the callee that the call is recorded if the
caller consents to the recording.
If you are in the U.S.,
Phil Finkler wrote:
No, I didn’t have m added. Should I have it added? I know I’ve ran Asterisk
with mp3123 in the past and music worked ok. It seems when I hit the hold
button on the phones, it does trigger the message saying music on hold is
starting but it INSTANTLY stops. I wish it
Lee Jenkins wrote:
I was wondering the same thing as my MOH isn't working either in a
1.2.14 installation so I'm recompiling mpg123 as per:
http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+redhat
We know you obviously need to use the m flag for the caller to hear
MOH when
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