Asterisk Queues Agents

Asterisk Asterisk Dialplan
1

Hello I use microsip and I would like to set up a queue. I assigned a dial plan to my users from my pjsip file. I also set up my file. Why is extension 1001 (user01) not ringing?
agents.conf

[support]
group=1
agent => 1001,1001,user01
agent => 1002,1002,user02
agent => 1003,1003,user03
agent => 1004,1004,user04

queues.conf

[support]
musicclass=default
stategy=rrmemory
joinempty=yes

member=>Local/1001@agents
member=>Local/1002@agents
member=>Local/1003@agents
member=>Local/1004@agents

member => Agent/@1

extensions.conf

[from-internal]

exten => 1001,1,Dial(PJSIP/user01,15,tTr)
exten => 1001,2,Voicemail(1001@boite-vocale)
exten => 1001,3,hangup

exten => 1002,1,Dial(PJSIP/user02,15,tTr)
exten => 1002,2,Voicemail(1002@boite-vocale)
exten => 1002,3,hangup

exten => 1003,1,Dial(PJSIP/user03,15,tTr)
exten => 1003,2,Voicemail(1003@boite-vocale)
exten => 1003,3,hangup

exten => 1004,1,Dial(PJSIP/user04,15,tTr)
exten => 1004,2,Voicemail(1004@boite-vocale)
exten => 1004,3,hangup

exten=>1005,1,VoiceMailMain(@boite-vocale)

[from-external]

include=> from-internal

exten => s,1,Answer()
exten => s,2,Queue(support)

2

You have no dialplan context called agents. That’s the basic reason.

Moreover, you are either using a version of Asterisk that is way past end of life, or you are using documentation for agents.conf that obsolete. Call back agents no longer exist, the syntax has changed, and I don’t think groups, and agent channels in general, are supported any longer.

3

What should I do then?

4

my context is from-internal in pjsip.conf and in extensions.conf

5

I don’t know how to use my dial plan in the queue.

I couldn’t find any up-to-date documentation

6

– Nobody picked up in 15000 ms
– Nobody picked up in 15000 ms
– Nobody picked up in 15000 ms
– Nobody picked up in 15000 ms
== Spawn extension (from-internal, 1002, 1) exited non-zero on ‘Local/1002@from-internal-00000027;2’
== Spawn extension (from-internal, 1003, 1) exited non-zero on ‘Local/1003@from-internal-00000026;2’
== Spawn extension (from-internal, 1001, 1) exited non-zero on ‘Local/1001@from-internal-00000028;2’
== Spawn extension (from-internal, 1004, 1) exited non-zero on ‘Local/1004@from-internal-00000029;2’
– Called Local/1003@from-internal
– Executing [1003@from-internal:1] Dial(“Local/1003@from-internal-0000002a;2”, “PJSIP/user03,15,tTr”) in new stack
– Called Local/1002@from-internal
– Called Local/1001@from-internal
– Called Local/1004@from-internal
– Called PJSIP/user03
– Executing [1002@from-internal:1] Dial(“Local/1002@from-internal-0000002b;2”, “PJSIP/user02,15,tTr”) in new stack
– Executing [1001@from-internal:1] Dial(“Local/1001@from-internal-0000002c;2”, “PJSIP/user01,15,tTr”) in new stack
– Executing [1004@from-internal:1] Dial(“Local/1004@from-internal-0000002d;2”, “PJSIP/user04,15,tTr”) in new stack
– Local/1003@from-internal-0000002a;1 is ringing
– Called PJSIP/user02
– Called PJSIP/user01
– Called PJSIP/user04
– Local/1004@from-internal-0000002d;1 is ringing
– Local/1001@from-internal-0000002c;1 is ringing
– Local/1002@from-internal-0000002b;1 is ringing
– Local/1003@from-internal-0000002a;1 connected line has changed. Saving it until answer for PJSIP/register-0000003c
– Local/1004@from-internal-0000002d;1 connected line has changed. Saving it until answer for PJSIP/register-0000003c
– Local/1001@from-internal-0000002c;1 connected line has changed. Saving it until answer for PJSIP/register-0000003c
– Local/1002@from-internal-0000002b;1 connected line has changed. Saving it until answer for PJSIP/register-0000003c

7

@jcolp Hello do you have any information about this?

8

Please do not tag people, including myself. If I have anything of value to add to a post then I will reply. This is a community forum, so there is no guarantee of timeframes on responses or if any will occur.

9

Local Channels

https://wiki.asterisk.org/wiki/display/AST/Local+Channel

Agent Pool, although I don’t think you need this:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_app_agent_pool
asterisk/app_agent_pool.c at master · asterisk/asterisk · GitHub (header comments)
asterisk/agents.conf.sample at master · asterisk/asterisk · GitHub

11

The problem with what you sent is that I already have users in my pjsip file: user01,user02,user03,user04. I have given numbers in the numbering plan in extensions.conf from there I added the numbers in queues.conf but it doesn’t work. Do you have the solution please? :slight_smile:

12

See the first link.

13

Hello my extensions are in PJSIP so the examples are not good for me.
I have to call my users who have names like “user01” for this I have created a numbering plan so that they call them for example 1001 for user01 the problem is that it does not work. I also need to create a dial plan for joining and leaving the queue. I sent you the configuration but it does not work.

14

Local channels are the way of making calls to other directory numbers where a device number is normally required, Your problem is that you don’t appear to understand how local channels are used. The first link is an explanation of how they are used.

My aim in answering is generally to result in people learning how things work, so that they can generalise to similar examples in the future, and understand why the solution goes wrong, if it does, not to provide material to be copied and pasted. It also means that I don’t have to spend significant time testing it. As such I will generally point out what is wrong and may point to existing documentation, but I’ll almost never give a complete worked solution.

15

I looked at my setup and my phone should ring but it doesn’t.

16

I used names instead of numbers for my users in pjsip.conf I think that’s the problem. What solution for that?

17

That is not a problem. I do that all the time, and it works fine.

Try reading these, to figure out how to dial PJSIP channels.

https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels

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

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

18

That is not your problem. Your problem is referencing a non-existent dialplan context.

You may have other problems, e.g. having voicemail answer a call on a queue, but there may be some strange reason why you want that behaviour, and I was only giving the simplest cause of the phone’s not ringing.

For the avoidance of doubt, endpoint names, SIP user names, and extension are not restricted to being numeric. The main thing that results in extensions normally being numeric is the difficulty of entering anything no-numeric on a normal phone.

19

I would like to associate in the dial plan a number like 1001 to user01

20

Voicemail is not working, either there may be a report

– Executing [1001@from-internal:1] Dial(“PJSIP/user01-000000a3”, “PJSIP/”) in new stack
[Jul 27 08:59:12] WARNING[28777][C-00000025]: app_dial.c:2605 dial_exec_full: Dial argument takes format (technology/resource)
== Spawn extension (from-internal, 1001, 1) exited non-zero on ‘PJSIP/user01-000000a3’

21

You do appear to have called a local channel successfully here, but the log is incomplete, and your supplied queue configuration and dialplan could not have got your here. There is no evidence of a queue being used in the log extract.

That log shows that all four devices reported they were ringing. If they weren’t actually ringing, that is a problem on the device, although using pjsip set logger on to double check that 180 Ringing actually came from the device would further confirm this.