Re: [asterisk-users] how to make asterisk set cos values

[hw]

On Friday, January 31, 2020 12:33:17 PM CET hw wrote:
> Hi,
> 
> examining the network traffic with wireshark shows that asterisk does not
> set any QoS values at all.
> 
> What do I need to do to make asterisk set QoS values (on Centos 7)?
> 
> The wiki says to use vconfig to set QoS values[1].  What does the
> skb-priority need to be set to?  How do you use vconfig on interfaces that
> are not VLAN interfaces?
> 
> Is it generally impossible to set QoS values on bonding interfaces?
> 
> 
> [1]: https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service

Any ideas?




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Re: [asterisk-users] Hangup hook to put back a call into a queue

[David P]

It might work for you to branch on ${DIALSTRING} just after your Dial
command, if you want to handle a BUSY, NOANSWER, or other result. But if
the peer of that Dial hungup, then based on what Joshua said, it seems
there's no recovery.
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Re: [asterisk-users] Hangup hook to put back a call into a queue

[Joshua C. Colp]

On Wed, Feb 5, 2020 at 12:34 PM Farkas Levente  wrote:

> hi,
> I hope someone can help me:-)
> we’ve got a freepbx server. there are 2 special extensions (2001, 2002).
> if someone calls this extensions (or a call is forwarded to these
> extensions) and these extension hangup (not the caller party), then we’d
> like to put the calls back into a queue (1000) and wouldn’t like to hangup.
>
> I read your description about hangup hooks:
> https://community.freepbx.org/t/hooking-for-fun-and-income/57718
>
> but still not able to implement it:-(
> what I’ve done:
> * found out in a hard way how to detect the current destination
> extension (because it’s turn out that CALLERID(dnid) is not working in
> case of forwarded call it’s show the original destination)
> * write a macro-dialout-one-predial-hook and a hook marco like this:
>
> [macro-dialout-one-predial-hook]
> exten => s,1,Noop(Entering user defined context
> macro-dialout-one-predial-hook in extensions_custom.conf)
> exten => s,n,GotoIf($["${DEXTEN}"=“2001”]?special)
> exten => s,n,GotoIf($["${DEXTEN}"=“2002”]?special)
> exten => s,n,MacroExit
> exten => s,n(special),NoOp(--- Push Special Hangup Handler
> --)
> exten => s,n,Set(CHANNEL(hangup_handler_push)=back-to-1000-hangup,s,1)
> exten => s,n,MacroExit
>
> [back-to-1000-hangup]
> exten => s,1,Noop(== Entering user defined context
> back-to-1000-hangup ===)
> exten => s,n,Queue(1000)
> exten => s,n,Return
>
> it seems to be called and seem to enter into to call but immediately
> hangup.
> first of all, in this case when in the hangup handler I will NOT like to
> hangup how should I finish the marco?:
>

Hangup handlers don't allow you any control over the hangup process. You
can't stop it from occurring and in fact when it occurs the channel is
already hung up. Anything that expects a live channel won't work.


> Hangup
> Return
> MacroExit
> how to redirect the call to the queue?:
>
> Queue(1000)
> ChannelRedirect(${CHANNEL},,1000,1)
> Gosub(ext-intercom,*801000,1())
> dial-one,HhTtrM(auto-blkvm),1000
> and what is the reason I can’t put the call back to the queue?
> I know that I'm already in the hangup sequence, but still wouldn't like
> to hangup.
> or this can't be done in the hangup handler?
>

You can't do it from a hangup handler. The Dial option provides the g
option[1] which can be used to continue dialplan execution when the called
party hangs up, but I don't work on FreePBX so I can't comment on how best
to use it there.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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[asterisk-users] Hangup hook to put back a call into a queue

[Farkas Levente]

hi,
I hope someone can help me:-)
we’ve got a freepbx server. there are 2 special extensions (2001, 2002). 
if someone calls this extensions (or a call is forwarded to these 
extensions) and these extension hangup (not the caller party), then we’d 
like to put the calls back into a queue (1000) and wouldn’t like to hangup.


I read your description about hangup hooks:
https://community.freepbx.org/t/hooking-for-fun-and-income/57718

but still not able to implement it:-(
what I’ve done:
* found out in a hard way how to detect the current destination 
extension (because it’s turn out that CALLERID(dnid) is not working in 
case of forwarded call it’s show the original destination)

* write a macro-dialout-one-predial-hook and a hook marco like this:

[macro-dialout-one-predial-hook]
exten => s,1,Noop(Entering user defined context 
macro-dialout-one-predial-hook in extensions_custom.conf)

exten => s,n,GotoIf($["${DEXTEN}"=“2001”]?special)
exten => s,n,GotoIf($["${DEXTEN}"=“2002”]?special)
exten => s,n,MacroExit
exten => s,n(special),NoOp(--- Push Special Hangup Handler 
--)

exten => s,n,Set(CHANNEL(hangup_handler_push)=back-to-1000-hangup,s,1)
exten => s,n,MacroExit

[back-to-1000-hangup]
exten => s,1,Noop(== Entering user defined context 
back-to-1000-hangup ===)

exten => s,n,Queue(1000)
exten => s,n,Return

it seems to be called and seem to enter into to call but immediately hangup.
first of all, in this case when in the hangup handler I will NOT like to 
hangup how should I finish the marco?:


Hangup
Return
MacroExit
how to redirect the call to the queue?:

Queue(1000)
ChannelRedirect(${CHANNEL},,1000,1)
Gosub(ext-intercom,*801000,1())
dial-one,HhTtrM(auto-blkvm),1000
and what is the reason I can’t put the call back to the queue?
I know that I'm already in the hangup sequence, but still wouldn't like 
to hangup.

or this can't be done in the hangup handler?

thank you for your help in advance.

regards.
--
  Levente   "Si vis pacem para bellum!"

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[asterisk-users] Always Be Conferencing v16e - pure AEL-based dial plan solution

[C.Maj]

/
 *  *
 *  Always Be Conferencing (ABC)*
 *  *
 *  Creator: chris @ Penguin PBX Solutions  *
 *  Version: 16e "Elevation Gain Edition"   *
 *  Release: 2020-02-04 *
 *  License: Creative Commons Zero v1.0 Universal (CC0 1.0) *
 *  *
 /

/*

Howdy
=

ABC starts your Conference Bridge in ASTERISK(R) by Originating calls to up
to twenty other participants, after you dial just one (configurable) number.

Your friends' phones will immediately be notified of your call attempt.
The caller ID information they see on their SIP phones will show them your name,
number, GPS co-ordinates (if configured), and the remote end destination number
you originally dialed (the DNID).

When your friends answer the phone, they will hear some short notes (A-B-C),
and then be able to talk with you -- even before the remote end answers!

In Simple Mode, the conference will continue until all parties disconnect
-- even if the original caller (you) hang up the phone before anybody answers!

ABC is tested most with PJSIP on ASTERISK certified version 16.3-cert1.
It should also work with any channel technology (see DAHDI in Example #2.)
Non-numeric extension numbers should work, but you should test it yourself.
Really, you should test and verify everything in ABC that you intend to use!



FEATURES


~ Instant (Simple Mode) or delayed (Tin Can Mode) bridging for caller.
~ Dynamic temporary pool call back number <-> caller ID mapping and routing.
~ DIY phone-based geolocation update support for changing GPS info via IVR.
~ Single press DTMF options to play back dispatachable location in-band.
~ Optional automated playback of GPS co-ordinates to remote end upon answer.
~ Preliminary support for RFC6442 Geolocation headers with some SIP providers.
~ Saves first thirty seconds of call as voicemail (to email) (if configured.)
~ Optional full call recording of each participant leg of the conference.
~ Easy integration with existing ASTERISK and FreePBX(R) installations.



Quick Start mini-HOWTO for first time ABC users on FreePBX
==

1. First, SSH into your PBX, and then run the following commands:

mkdir ~/abc
cd ~/abc
wget 
https://raw.githubusercontent.com/chrsmj/always-be-conferencing/master/extensions-pngnpbx-abc-v16e.ael2
echo '#include "extensions-pngnpbx-abc-v16e.ael2"' > extensions.ael
echo '[from-internal-custom]' > extensions_custom.conf
echo 'exten => _123*222,1,Goto(pngnpbx-abc-demo,${EXTEN},1)' >> 
extensions_custom.conf
echo 'exten => _123*222X*.,1,Goto(pngnpbx-abc-demo,${EXTEN},1)' >> 
extensions_custom.conf

2. Now you are about to copy the files into the Asterisk configuration 
directory.
   You will probably need to be root or use sudo access to complete this next 
step:

sudo cp -i ~/abc/extensions* /etc/asterisk/

   If there are any errors on the copying, then you should manually inspect
   the files for conflicts. Most likely, the only conflict will be with the
   extensions_custom.conf files. Run this command to check your existing file:

cat /etc/asterisk/extensions_custom.conf

   If it is empty, then you can try the copy again, and just clobber the old 
file:

sudo cp ~/abc/extensions_custom.conf /etc/asterisk/

   If the existing file is not empty, then you will need to manually insert the
   ABC "_123*222..." config lines into the existing [from-internal-custom] 
context
   (or create the context if it does not already exist.)

3. After completing the file copying, exit SSH and open up your PBX GUI.
   Issue a full PBX reload using your GUI (sometimes you have to make a small
   change to get the "Apply Changes" button to appear, so make a trivial thing 
like
   adding ABC after your name, then "Submit", then "Apply Changes".)

4. Finally, start making test calls from your desk phone, with massive dial 
strings:

123*2224*1000*2000*3000

   ...replacing 1000 with an extension acting as your "remote end" operator
   and replacing 2000 and 3000 with your own local "friend" extensions.

   To hear your GPS co-ordinates, any participant can press * at any time
   during the call. To stop the bot playback, press #.

   To silence the ring-back you hear from friend's phones, but not the remote 
end,
   you can run the following command on the Asterisk CLI:

localhost*CLI> indication add us silent 0/15000


Note this mini-HOWTO assumes all phones are using the same channe

Re: [asterisk-users] Asterisk 13.31.0 and 16.8.0 - Bridge problem on incoming calls

[Joshua C. Colp]

On Tue, Feb 4, 2020 at 3:11 PM Administrator  wrote:

> Done, ASTERISK-28725
>  but against
> version 16,7.0 and 13.30.0 as the new ones aren't available in the list
>
> For those interested this has been resolved. The patch files were used and
"make clean" was not run. The compiler did not rebuild everything resulting
in some of Asterisk being the old version and some being the new version.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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