If there is no phone ringing (because all operators are busy and the call
is still waiting in the queue) then I cannot pickup the call with
Pickup(Queuename@PICKUPMARK):
app_directed_pickup.c:302 pickup_exec: No target channel found for
magazzino@PICKUPMARK.
Any idea?
Niccolo'
On lunedì 31
On lunedì 31 ottobre 2016 20:16:30 CET, Freddi Hansen wrote:
you could use the PICKUPMARK with the Pickup().
before you call the Queue app you set PICKUPMARK=Queuename.
When you want to pickup the call you do
Pickup(Queuename@PICKUPMARK) to only get calls in the Queue with
Queuename.
That
Hi,
I'm currently using Pickup() to pickup calls from queues, but this is VERY
annoying because often users from different queues dialed the very same
extension (for example they pressed '1' to reach something but in different
submenus). Other times they didn't dial anything but they end up in
Hi,
I think almost everybody did it once: you call the pbx, the pbx hangs out, it
calls back using the caller id, you type the number you want to dial using
dtmf and the pbx calls the number for you. Such a way you can make a
completely free call. Of course you should implement CLI whitelists
Hi,
I have a Portech MV-374 GSM Gateway and I'd like to send SMS from a web
page to confirm the subscriptions. How can I achieve it? Is Asterisk of
any use to send SMS with the Portech? I really have no idea because I
know nothing about the whole SMS thing...
Thanks,
Niccolò
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Il 22/10/2012 18:44, Alex Forster ha scritto:
*DEVELOPERS* - If I took a crack at fixing this issue, what general tips do
you have for me to make it most likely that my solution can be integrated
into HEAD? I believe I can justify spending some time at work to deal with
this, but not without at
-answer-with-callwaitingyes-on-fxs-channels.html
Il 09/10/2012 13:34, Niccolò Belli ha scritto:
Hi,
I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a
remote peer and an fxs phone gets connected and the remote peer hangsup,
then asterisk sends the Idle state to notify the watcher
-fxs-idle-hints.html
Il 08/10/2012 13:20, Niccolò Belli ha scritto:
I will make an example:
A is an fxs phone with callwaiting=yes in chan_dahdi.conf
X calls A. A answers.
Y calls A. A hears the call waiting tone.
Now if A hangs up before X, then A rings again (which is what I want).
BUT if X hangs
Hi,
I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a
remote peer and an fxs phone gets connected and the remote peer hangsup,
then asterisk sends the Idle state to notify the watcher before you
hangup the fxs phone! Such a way if the user forgets to hangup the fxs
phone
Il 09/10/2012 13:34, Niccolò Belli ha scritto:
Hi,
I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a
remote peer and an fxs phone gets connected and the remote peer hangsup,
then asterisk sends the Idle state to notify the watcher before you
hangup the fxs phone! Such a way
Il 09.10.2012 21:24 Mike Diehl ha scritto:
I hope no one considers this off topic...
I have a phone customer who wants 2 Internet connections so that if
one goes down, he can use the other for phone service.
So, I'd like to get a recommendation for a relatively inexpensive
router that can
Il 09.10.2012 23:04 James Sharp ha scritto:
Do you have your phones set for a short register time? Otherwise the
far end might have stale contact information to send incoming calls
back to.
Actually I use the failover only for the nat clients, my pbx has a
public ip on the interface and it
I will make an example:
A is an fxs phone with callwaiting=yes in chan_dahdi.conf
X calls A. A answers.
Y calls A. A hears the call waiting tone.
Now if A hangs up before X, then A rings again (which is what I want).
BUT if X hangs up first, then A automatically answers Y without even
ringing.
Hi,
Il 02/10/2012 21:04, Mc GRATH Ricardo ha scritto:
How about to change tone list on indications.conf file?
As I already said indications.conf doesn't work for dahdi channels,
unfortunately the callwaiting tone is hardcoded in asterisk itself (not
even in dahdi/libtonezone!). I solved
Thank you, I already considered such an approach but the customer wanted
to receive the new call *immediately* after the hangup (basically
because it was possible with the old pbx).
This is how I solved: http://www.spinics.net/lists/asterisk/msg153399.html
Such a way I hear the annoying beep
Hi,
The call waiting tone is very annoying (you hear nothing while it plays
the beep). I need callwaiting because of the queues (the phone has to
ring as soon as you hangup) but I want to remove the beep on my dahdi
channels, how can I do?
Thanks,
Niccolò
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Il 01/10/2012 17:12, Danny Nicholas ha scritto:
I would start here
http://www.voip-info.org/wiki/view/Asterisk+indications+default
You could change the tone to something less annoying or just inaudible.
Does it affect dahdi channels? If I recall correctly the dahdi tones are
Il 01/10/2012 17:47, Danny Nicholas ha scritto:
Maybe /etc/asterisk/chan_dahdi.conf
No, the only option here is to enable/disable callwaiting.
Cheers,
Niccolò
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Is it hardcoded in zonedata.c, am I right?
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I modified the Italian zone in zonedata.c from
{ DAHDI_TONE_CALLWAIT, 425/400,0/100,425/250,0/100,425/150,0/14000 },
to
{ DAHDI_TONE_CALLWAIT, 0/14 },
but I can still hear the damn beep :(
I even rebooted the pc, suggestions?
Niccolò
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Il 01/10/2012 20:08, Danny Nicholas ha scritto:
This is probably a dumb question, but your country/zone is set to it
(installs as us by default)?
Obviously :)
Anyway I think I found where the fucking bastard is hardcoded:
chan_dahdi.c in asterisk :)
I will change
#define
Il 26/04/2012 16:04, Mark Michelson ha scritto:
What is the strategy of the queue?
Ringall.
How are the queue members listed (i.e. are they SIP channels or
local channels)?
There is only one member listed: SIP/phone-200
My suspicion is that the queue is simultaneously
dialing local
Hi,
Il 18/04/2012 00:39, Kevin P. Fleming ha scritto:
You guys know that it works in Asterisk 10, but you say you can't use
Asterisk 10 for some reason that I don't understand.
1) No Debian packages for v10. If you have to maintain lots of servers,
installing from sources is a big burden.
Il 18/04/2012 14:50, Kevin P. Fleming ha scritto:
Do you expect Debian-style packages to include these third-party
components in Asterisk? If you are talking about DAHDI specifically,
moving to Asterisk 10 does not change DAHDI requirements at all.
No, I just pointed out that upgrading to a
Il 18/04/2012 14:50, Kevin P. Fleming ha scritto:
we'll get this corrected
That's an awesome news indeed.
Niccolò
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Il 17/04/2012 01:10, Niccolò Belli ha scritto:
Tomorrow I will try without directmedia=yes.
Unfortunately it didn't help.
Niccolò
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New to Asterisk
Patch:
https://issues.asterisk.org/jira/secure/attachment/42605/local_remote_hint2.diff
https://issues.asterisk.org/jira/browse/ASTERISK-16735?focusedCommentId=191720page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-191720
It's already merged in asterisk 10.4-rc1,
I suspected it, but it didn't work at first. I fear I didn't understand
what the context refers to in Pickup(extension[@context]).
I will make an example: phone-100 wants to pick up a ringing phone-200
(call comes from my-sip-provider).
This is my sip.conf
[phone-100]
context=context-100
Hi,
Il 16/04/2012 22:50, Larry Moore ha scritto:
Do you have directmedia=no in your SIP configuration?
Yes I have.
Niccolò
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Il 20/01/2012 20:32, Alec Davis ha scritto:
This maybe not what you want.
Our solution was monitor a queue with a BLF, instead of a queue member
This reviewhttps://reviewboard.asterisk.org/r/1619/ allows a BLF lamp to
flash when a queue is ringing, then the queue can be picked up by the BLF
https://issues.asterisk.org/jira/browse/ASTERISK-19684
Can someone help me? It's an old problem I have since the earlier
patches which isn't still solved. I'd like to use T.38 gw in production...
Cheers,
Niccolò
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Hi,
If someone is interested I made Debian Squeeze Packages:
http://www.linuxsystems.it/2012/04/asterisk-1-8-11-0-debian-squeeze-packages-with-t-38-gateway-queue-hints-and-fixed-rfc4235/
Niccolò
Il 30/03/2012 17:22, Niccolò Belli ha scritto:
http://www.linuxsystems.it/2012/03/new-t-38-gateway
http://www.linuxsystems.it/2012/03/new-t-38-gateway-patch-against-asterisk-1-8-11-0/
I made a new patch from irroot's branch and I ported it to 1.8.11.
Unfortunately latest one is still against 1.8.8 and porting from
subversion is quite time consuming, hopefully my work will be useful to
Il 30/03/2012 19:29, Ryan Wagoner ha scritto:
It looks like the patch is a backport of the t.38 gateway functionality
in Asterisk 1.10.
Yes it's a backport from asterisk 10, asterisk 1.8 does not have the t38
gateway functionality and there is no chance to get t38 gw in 1.8 at
this point.
Il 30/03/2012 19:16, Bryant Zimmerman ha scritto:
Why is it not in Jarr?
Ok I linked it in jira, anyway I don't know how many peoples still
follow the old bug report considering it has been closed.
Niccolò
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I noticed it by chance while digging into the code, I think it's a great
news!
Darkbasic
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Hi,
Is it possible? I tried AddQueueMember(SIP/$TRUNK/number) but it tries
to call SIP/$TRUNK instead.
Cheers,
Darkbasic
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Il 31/01/2012 15:42, C F ha scritto:
Use local channel
Thanks, I completely forget about local channel.
Darkbasic
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Il 31/01/2012 15:46, Doug Lytle ha scritto:
You'll also want to keep track of the number of active calls, since, I
believe, the queue app will not be able to see signaling on that line.
I'm sorry but, how to?
Darkbasic
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Il 25/01/2012 22:52, Michael Keuter ha scritto:
Outcry! :-)
I'm outcrying too :)
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Il 23/01/2012 21:03, Olivier ha scritto:
How can I test this solution on a 1.8.8.1 system ?
If I'm not mistaken, diffhttps://reviewboard.asterisk.org/r/1619 do
not apply to 1.8.8.1.
Are you sure? Hopefully I will test it in the week end.
Darkbasic
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Hi,
I have some phones monitoring several extensions, I want them being able
to pickup calls using Busy Lamp Field. Unfortunately it doesn't work
when the calls come from a queue.
Example:
Phone 110 wants to monitor phone 102. Phone 102 is a member of the queue
Test, it has been added to
Hi,
I did map a key in each phone to add it to the incoming call queue
(using AddQueueMember). It also updates a custom hint state for the Busy
Lamp Field (BLF).
When I restart asterisk it keeps the previous Hint states (I don't know
how, but it does), but obviously the phone is no longer a
Hi,
I set verbose to 3, but I do not see any RINGING notification in the
CLI. On the contrary, when the phone goes UNREACHABLE I get:
[Dec 13 21:10:06] NOTICE[9988]: chan_sip.c:25533 sip_poke_noanswer: Peer
'152' is now UNREACHABLE! Last qualify: 130
== Extension Changed 152[blf] new state
Il 19/04/2011 23:41, Kevin P. Fleming ha scritto:
If you are the receiver of the call (and thus they are the sender of the
call), it is *your* system's responsibility to initiate the switch to
T.38, not theirs.
Are you sure? So what's faxdetect=t38 for?
Cheers,
Darkbasic
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Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto:
Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and
internet is offline.
srvlookup = no didn't help.
What about putting my provider's name in /etc/hosts?
Should it solve the problem?
A caching nameserver is not a
Hi,
this is an old outstanding problem, unfortunately I don't remember how
to walkaround it. I use asterisk 1.8.3 and I have a public IP in my
network interface. As soon as the Internet connection goes down, phones
stop working. I want to be able to use pstn, isdn and the gsm gateway
even if the
Hi, I'd like to replace my DECT + fxs phones with some wireless phones.
Main problem is: what about the roaming from one ap to another? Do
someone adopted the 802.11r standard? What APs and what phones do you
suggest me? I'm open to suggestions but I'd like to avoid proprietary
solutions, I don't
Il 14/04/2011 12:25, Larry Moore ha scritto:
I made a suggestion on how you could check this i.e. have your incoming
call go directly to the fax extension, my 1.8.3.2 installation
immediately negotiates a T.38 connection in this sceanrio, of course I
enabled the fallback option so it will use
Il 14/04/2011 14:34, Larry Moore ha scritto:
allow=alaw,g729 ; alaw required for T.30 facsimile if T.38 fails to
I didn't understand the point. If you enable both alaw and g729 it will
simply use the preferred one: if it's g729 tone based detection will
still not work, if it's alaw I will still
Hi, I continue the discussion from
https://issues.asterisk.org/view.php?id=19103
If T.38 reinvite detection should still work, why it doesn't?
If I use faxdetect = t38 it does never detect the fax, even using alaw.
Cheers,
Darkbasic
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Il 13/04/2011 19:54, Larry Moore ha scritto:
That is because the remote endpoint, eutelia, will need to detect the
Fax Tones and send the T.38 ReINVITE to you, they may not have T.38
enabled.
Uhm... it's very unlikely.
As a suggestion you could configure your incoming calls from eutelia to
Hi, when I receive a call from ISDN BRI (with a Sangoma A500) and I try
to playback something I get the following error:
**[WOOMERA]** HW DTMF supported s1c1-
-- Executing [number@from-pstn:1] Answer(WOOMERA/g1/1-7b29, ) in
new stack
**[WOOMERA]** +++ANSWER WOOMERA/g1/1-7b29
-- Executing
What about transparent t.38 gatewaying? :-(
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I do not use windows on the desktop/laptop, but when I have to I use putty.
Darkbasic
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Awesome!
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No one can help me?
Darkbasic
Il 18 ottobre 2009 15.49, Niccolò Belli darkbas...@gmail.com ha scritto:
2009/10/17 Paul Hales pdha...@optusnet.com.au:
I have used the group function to limit the calls entering a queue for a
similar reason to yourself.
But I do not want to limit the calls
Hi, I already considered such a solution: I will have lots of loops
and the customer will loose its position if someone else who called
later will find the queue 'ready to join' before him.
I will also have a music on hold mismatch when he will join the queue
and I will loose all the benefits of
2009/10/17 Paul Hales pdha...@optusnet.com.au:
I have used the group function to limit the calls entering a queue for a
similar reason to yourself.
But I do not want to limit the calls entering a queue (I can already
do it with maxlen= in queues.conf), people should wait in the queue
until an
Hi,
I explain what I want to do..
All the operators share their phones. The number of the operator isn't
constant, so it's possible that two operators share all the phones.
They need to move all around, so they pick up the first phone they find.
If there are only few operator is very annoying for
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