Hotline - play message in Queue/Dial before connect call

tuxmartin · January 1, 2023, 9:08pm

Hi, I need to create a hotline for my customers.
Once the customer calls, Asterisk hands the call over to the first technician. If he is not available, then the next one. Each customer has their own PIN. Without PIN, there will be no connection to the technician.
That works for me.

After connecting the call to the technician, I want to play the text with the customer’s name/PIN (so that the technician knows who is calling) and then connect the call. But, unfortunately, I don’t know how to do it.

My config:

# -------------------------------------------
# cat /etc/asterisk/queues.conf
[general]
persistentmembers = yes
monitor-type = MixMonitor

[queue_template](!)
musicclass=company
strategy=linear
joinempty=yes
leavewhenempty=no
ringinuse=no

[techsupport](queue_template)
member = SIP/00420111111111@trunk
member = SIP/00420222222222@trunk

# -------------------------------------------
# cat /etc/asterisk/extensions.conf
[incomming]
exten => s,1,MixMonitor("/var/log/hotline/${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}_${CALLERID(num)}.wav")
exten => s,n,Playback(hello)
exten => s,n,DBdeltree(customer)
exten => s,n,Set(DB(customer/12345)=1)
exten => s,n,Set(DB(customer/54321)=1)
exten => s,n,Playback(conf-getpin)
exten => s,n,Read(CUSTOMER,,0,,1,60)
exten => s,n,GotoIf(${DB_EXISTS(customer/${CUSTOMER})}?:customer,s,100)
exten => s,n,Queue(techsupport)
exten => s,n,Playback(goodbye)
exten => s,n,Hangup
exten => s,100,Playback(conf-invalidpin)
exten => s,101,Hangup

wusel · January 1, 2023, 10:08pm

I used this for a while:

; extensions.conf
[macro-announce-to-queue-member]
exten => s,1,NoOp(announce-to-queue-member)
same  => n,Wait(0.5)
same  => n,Flite("Time in Queue: ${QEHOLDTIME} seconds.")
same  => n,MacroExit

Along with:

; queues.conf
[my-queue]
musicclass=ours
joinempty=no
leavewhenempty=no
ringinuse=no
reportholdtime = no
strategy=ringall	
... more stuff ...
membermacro=announce-to-queue-member
setqueuevar=yes
setqueueentryvar=yes
... members ...

This turned out to be more irritating than useful:

caller has to wait in silence while the text is being spoken to the agent
agent’s habit is to answer the phone stating name and company, not to await such a message
times are usually too short to warrant such an info anyway

But your mileage may, of course, vary

tuxmartin · January 2, 2023, 10:08am

@wusel Hello, it works great. Thanks.

You were right. That silence sounds strange.
Can I keep playing musicclass to the caller until the membermacro finishes?

PitzKey · January 2, 2023, 10:39am

Hi @tuxmartin,

Macro() has been deprecated. Instead, use membergosub [1]
IIRC, if you use the announce option, it will not stop MOH, but I may be wrong on that. [2] [3]

[1]

github.com

asterisk/asterisk/blob/master/configs/samples/queues.conf.sample#L267


      
          ; if set, run this macro when connected to the queue member
          ; you can override this macro by setting the macro option on
          ; the queue application
          ;
          ;membermacro=macro_name[,arg1[,...][,argN]]
          
          
; if set, run this gosub when connected to the queue member
          ; you can override this gosub by setting the gosub option on
          ; the queue application
          ;
          ;membergosub=gosub_context_name[,arg1[,...][,argN]]
          
          
; How often to announce queue position and/or estimated
          ; holdtime to caller (0=off)
          ; Note that this value is ignored if the caller's queue
          ; position has changed (see min-announce-frequency)
          ;
          ;announce-frequency = 90
          ;
          ; The absolute minimum time between the start of each
          ; queue position and/or estimated holdtime announcement

[2]
Set the announcement in the queue config:

github.com

asterisk/asterisk/blob/master/configs/samples/queues.conf.sample#L79


      
          ; dialplan (the latter of which overrides everything).
          ;
          ;musicclass = default
          ;
          ; An announcement may be specified which is played for the member as
          ; soon as they answer a call, typically to indicate to them which queue
          ; this call should be answered as, so that agents or members who are
          ; listening to more than one queue can differentiated how they should
          ; engage the customer
          ;
          ;announce = queue-markq
          ;
          ; A strategy may be specified.  Valid strategies include:
          ;
          ; ringall - ring all available channels until one answers (default)
          ; leastrecent - ring interface which was least recently hung up by this queue
          ; fewestcalls - ring the one with fewest completed calls from this queue
          ; random - ring random interface
          ; rrmemory - round robin with memory, remember where we left off last ring pass
          ; rrordered - same as rrmemory, except the queue member order from config file
          ;             is preserved

[3]
Pass the announcement dynamically

exten => s,n,Queue(techsupport,,,custom/${customer_name})

https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Application_Queue

tuxmartin · January 2, 2023, 3:44pm

Hi @PitzKey it works great:

exten => s,n,Queue(techsupport,,,/etc/asterisk/sounds/incoming-call-from)

Is it possible to play two files in Queue?

In the documentation, I found the following (Home - Asterisk Documentation):

announceoverride
filename - Announcement file(s) to play to agent before bridging call, overriding the announcement(s) configured in queues.conf, if any.
filename2[,filename2…]

But it doesn’t work for me:

exten => s,n,Set(GLOBAL(CUSTOMER_NAME)=test)
exten => s,n,Queue(techsupport,,,/etc/asterisk/sounds/incomming-call-from,/etc/asterisk/sounds/customer-${CUSTOMER_NAME})

david551 · January 2, 2023, 4:09pm

The structured comments actually say to use & not comma. It looks like the help generator is broken, or the markup syntax is wrong. The code treats the value exactly the same as announce in queues.conf, and, although I couldn’t find anything that allows multiple file names there, it is possible. NB, only enough space is reserved for one long path name.

			<parameter name="announceoverride" argsep="&amp;">
				<argument name="filename" required="true">
					<para>Announcement file(s) to play to agent before bridging call, overriding the announcement(s)
					configured in <filename>queues.conf</filename>, if any.</para>
				</argument>
				<argument name="filename2" multiple="true" />
			</parameter>

tuxmartin · January 2, 2023, 5:53pm

@david551

I changed it to

exten => s,n,Queue(techsupport,,,/etc/asterisk/sounds/incomming-call-from&/etc/asterisk/sounds/customer-${CUSTOMER_NAME})

and it still doesn’t work.

david551 · January 2, 2023, 6:37pm

Which version are you using? This first appeared in: 16.20.0, 18.6.0, 19.0.0, and 20.0.0.

See:

https://issues.asterisk.org/jira/browse/ASTERISK-29528

Note that versions 16 and 19 are in security fixes only state.

tuxmartin · January 2, 2023, 9:04pm

@david551 I have Asterisk 16.16.1. Which version do I need?

david551 · January 2, 2023, 10:41pm

The minimum sub-versions were listed in my previous posting. 16.20.0 for Asterisk 16.

If you have any Asterisk 16, you should be upgrading, probably to Asterisk 20.1.0.

tuxmartin · January 3, 2023, 12:13pm

@david551 I used Asterisk 18.15.0-r0 (Alpine Linux v3.17), and it works great. Thank you.

PitzKey · January 3, 2023, 12:13pm

Tested with 18.15.1

same => n,Queue(4110,,,SIP_Test_Failure&SIP_Test_Success)

Log:

[2023-01-03 07:09:21] VERBOSE[56180][C-00000006] pbx.c: Executing [*8803@from-internal:7] Queue("PJSIP/10111-00000006", "4110,,,SIP_Test_Failure&SIP_Test_Success") in new stack
[2023-01-03 07:09:21] VERBOSE[56180][C-00000006] res_musiconhold.c: Started music on hold, class 'default', on channel 'PJSIP/10111-00000006'
[2023-01-03 07:09:21] VERBOSE[56180][C-00000006] app_queue.c: Called PJSIP/2547
[2023-01-03 07:09:21] VERBOSE[54538] netsock2.c: Using SIP RTP Audio TOS bits 184
[2023-01-03 07:09:21] VERBOSE[54538] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2023-01-03 07:09:21] VERBOSE[54538] netsock2.c: Using SIP RTP Audio CoS mark 5
[2023-01-03 07:09:21] VERBOSE[56180][C-00000006] app_queue.c: PJSIP/2547-00000007 connected line has changed. Saving it until answer for PJSIP/10111-00000006
[2023-01-03 07:09:22] VERBOSE[56180][C-00000006] app_queue.c: PJSIP/2547-00000007 is ringing
[2023-01-03 07:09:26] VERBOSE[54538] res_rtp_asterisk.c: 0x7f479807f780 -- Strict RTP learning after remote address set to: XX.XX.XXX.XX:17549
[2023-01-03 07:09:26] VERBOSE[56180][C-00000006] app_queue.c: PJSIP/2547-00000007 answered PJSIP/10111-00000006
[2023-01-03 07:09:26] VERBOSE[56180][C-00000006] file.c: <PJSIP/2547-00000007> Playing 'SIP_Test_Failure.ulaw' (language 'en')
[2023-01-03 07:09:28] VERBOSE[56180][C-00000006] file.c: <PJSIP/2547-00000007> Playing 'SIP_Test_Success.ulaw' (language 'en')
[2023-01-03 07:09:39] VERBOSE[56180][C-00000006] res_musiconhold.c: Stopped music on hold on PJSIP/10111-00000006
[2023-01-03 07:09:39] VERBOSE[56251][C-00000006] bridge_channel.c: Channel PJSIP/2547-00000007 joined 'simple_bridge' basic-bridge <e636b45d-c16d-402b-86ff-0c0f562f4760>
[2023-01-03 07:09:39] VERBOSE[56180][C-00000006] bridge_channel.c: Channel PJSIP/10111-00000006 joined 'simple_bridge' basic-bridge <e636b45d-c16d-402b-86ff-0c0f562f4760>

As you can see, it plays fine and MOH continues to play.

Edit - jinks

system · February 2, 2023, 12:13pm

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Hotline - play message in Queue/Dial before connect call

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