Asterick is not connecting caller to agent after first call

Hello All,

I’m having an interesting issue with my Asterisk 13 setup that I don’t have the knowledge to debug. I have an asterisk server that receives calls from Twilio via their SIP trunking service. The asterisk box has two queues and I’m adding members to the queue dynamically in my dialplan. The agent calls into asterisk on a softphone and successfully gets a call from the queues on the first call made. On any subsequent call, the new caller is stuck in the queue even though my queue member is listed as no in use. I can repair the setup by un-registering and re-registering the softphone as a SIP peer and then calling back into the twilio number. Does anyone have any ideas on what might be wrong or some tips on debugging this? Thanks.

8dca4d752649*CLI> queue show
support has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s
   Members: 

      SIP/test-agent (ringinuse disabled) (dynamic) (Not in use) has taken no calls yet
   Callers: 
      1. SIP/twilio1-00000003 (wait: 0:06, prio: 0)


sales has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   No Members
   No Callers

Here is my queues.conf:

queues.conf
--------------

[general]
autofill=yes             ; distribute all waiting callers to available members
shared_lastcall=yes      ; respect the wrapup time for members logged into more 
                         ; than one queue

[StandardQueue](!)       ; template to provide common features
musicclass=default       ; play [default] music
strategy=ringall
joinempty=no             ; do not join the queue when no members available
leavewhenempty=yes       ; leave the queue when no members available
ringinuse=no             ; don't ring members when already InUse (prevents 
                         ; multiple calls to an agent)

[sales](StandardQueue)   ; create the sales queue using the parameters in the
                         ; StandardQueue template

[support](StandardQueue) ; create the support queue using the parameters in the
                         ; StandardQueue template

When the above issue happen , could you run sip show peers. Because based on this statement

it seems phones are losing the registration

Cheers for the reply, ambiorixg12.

Here’s the output before the first call is made:

8dca4d752649*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      

test-agent/test-agent     <ip>                             D  Auto (No)  No             32287    Unmonitored                                  
twilio1                   <ip>                                Auto (No)  No             5060     Unmonitored                                  

twilio2                   <ip>                                 Auto (No)  No             5060     Unmonitored                                  
twilio3                   <ip>                                 Auto (No)  No             5060     Unmonitored                                  
twilio4                   <ip>                                 Auto (No)  No             5060     Unmonitored                                  
5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 5 online, 0 offline]

It remains the same after the first call is complete and before the second, failing, call is made

Status is Unmonitored, please enable the qualify option

The peer is now monitored but unfortunately I’m still not getting the caller to leave the queue.

4ae4f2aaab62*CLI> queue show
support has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:3, SL:0.0% within 0s
   Members: 

      SIP/test-agent (ringinuse disabled) (dynamic) (Not in use) has taken no calls yet
   Callers: 
      1. SIP/twilio4-00000005 (wait: 0:03, prio: 0)


sales has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
   No Members
   No Callers


4ae4f2aaab62*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      

test-agent/test-agent    <ip>                             D  Auto (No)  No             32287    OK (109 ms)                                  

twilio1                   <ip>                                 Auto (No)  No             5060     Unmonitored                                  
twilio2                   <ip>                                 Auto (No)  No             5060     Unmonitored                                  
twilio3                   <ip>                                 Auto (No)  No             5060     Unmonitored                                  

twilio4                   <ip>                                 Auto (No)  No             5060     Unmonitored                                  
5 sip peers [Monitored: 1 online, 0 offline Unmonitored: 4 online, 0 offline]

show your queues.conf file

make sure autofill=yes

My queues.conf is in my OP and autofill is set to yes.

I don’t understand the OP part.

Sorry, original post. The first post.

I had the same problem with 13.8. Had to roll back to 13.5. Since then, there have been several patches that have fixed this and other problems (for me, at least). If you can, I’d recommend upgrading to 13.12.0-rc1.

There is a bug related to this issue, but it is for older version of Asterisk < 13

https://issues.asterisk.org/jira/browse/ASTERISK-20690

Ah, ok. I am also running 13.8. I’ll give the update or downgrade a try. Thanks a lot for the help!

FYI…There’s a problem with deadlocks in 13.5 when using chan_sip and realtime voicemail. Keep an eye on memory usage and taskprocessors. I ended up downloading the current v13 from git and applied the patch https://gerrit.asterisk.org/#/c/3962/4. It fixed all of my problems. I’m going to test the rc 13.12, and likely use that version once it’s released.

For expediency, I’ve downgraded to Asterisk 11 and everything I need to be working seems to be working well. Thanks very much for the help.

You should try upgrading no dowgrading Asterisk 11 is only days away from entering security fix only mode
give a try to the Asterisk 13.12.0- or > higher

Asterisk 11 Now Security Fix Only