What is the command to get the report?
“PJSIP/” only contains the technology, PJSIP, and not a resource (endpoint name and possibly other information). Again this couldn’t happen with the dial plan you have provided. It would generally happen because you were using a channel variable for the resource and gave the wrong name, or it was unset. There are no channel variable references in the configuration you supplied above.
Which file do you need to see to identify the problem?
am I accumulating problems? 
Full log with verbosity 5, and after issuing “pjsip set logger on” on the console, queues.conf. extensions.conf, and pjsip.conf. Also a description of the call(s) you made in the period covered by the log.
sent privately
I send you this message because I do not know if you have been notified for my private message
Sorry. I don’t look at things privately. You will generally find this is true.
See also:
http://www.catb.org/~esr/faqs/smart-questions.html#noprivate
When I say private it’s just the number and the users are present but not the passwords.
So you can watch.
I changed my configuration but my hotline phone still does not ring.
Connected to Asterisk 18.13.0 currently running on asterisk (pid = 920)
– Executing [s@from-external:1] Ringing(“PJSIP/register-0000001c”, “1”) in new stack
– Executing [s@from-external:2] Answer(“PJSIP/register-0000001c”, “”) in new stack
– Executing [s@from-external:3] Goto(“PJSIP/register-0000001c”, “ivr,s,1”) in new stack
– Goto (ivr,s,1)
– Executing [s@ivr:1] Answer(“PJSIP/register-0000001c”, “”) in new stack
– Executing [s@ivr:2] Set(“PJSIP/register-0000001c”, “TIMEOUT(response)=8”) in new stack
– Response timeout set to 8.000
– Executing [s@ivr:3] Queue(“PJSIP/register-0000001c”, “support”) in new stack
– Started music on hold, class ‘default’, on channel ‘PJSIP/register-0000001c’
– Called PJSIP/user01
– Executing [10@from-internal:1] Goto(“PJSIP/user01-0000001e”, “agent-login,s,1”) in new stack
– Goto (agent-login,s,1)
– Executing [s@agent-login:1] Answer(“PJSIP/user01-0000001e”, “”) in new stack
– Executing [s@agent-login:2] AddQueueMember(“PJSIP/user01-0000001e”, “support”) in new stack
[Jul 27 19:50:36] WARNING[14512][C-00000017]: app_queue.c:8346 aqm_exec: Unable to add interface ‘PJSIP/user01’ to queue ‘support’: Already there
– Executing [s@agent-login:3] Playback(“PJSIP/user01-0000001e”, “agent-loginok”) in new stack
– <PJSIP/user01-0000001e> Playing ‘agent-loginok.ulaw’ (language ‘fr’)
– Executing [s@agent-login:4] Hangup(“PJSIP/user01-0000001e”, “”) in new stack
== Spawn extension (agent-login, s, 4) exited non-zero on ‘PJSIP/user01-0000001e’
– Nobody picked up in 15000 ms
– Called PJSIP/user02
– Nobody picked up in 15000 ms
– Stopped music on hold on PJSIP/register-0000001c
== Spawn extension (ivr, s, 3) exited non-zero on ‘PJSIP/register-0000001c’
extensions.conf
[from-internal]
exten => 10,1,Goto(agent-login,s,1)
exten => 20,1,Goto(agent-logout,s,1)
exten => 1001,1,Dial(PJSIP/${user01})
exten => 1001,2,Voicemail(1001@boite-vocale)
exten => 1001,3,hangup
exten => 1002,1,Dial(PJSIP/${user02})
exten => 1002,2,Voicemail(1002@boite-vocale)
exten => 1002,3,hangup
exten => 1003,1,Dial(PJSIP/${user03})
exten => 1003,2,Voicemail(1003@boite-vocale)
exten => 1003,3,hangup
exten => 1004,1,Dial(PJSIP/user04,15,tTr)
exten => 1004,2,Voicemail(1004@boite-vocale)
exten => 1004,3,hangup
exten => 888,1,VoiceMailMain(s${CALLERID(num)})
[from-external]
exten => s,1,Ringing(1)
exten => s,2,Answer
exten => s,3,Goto(hotline,s,1)
exten => s,4,Hangup(16)
[agent-login]
exten => s,1,Answer()
exten => s,n,AddQueueMember(support)
exten => s,n,Playback(agent-loginok)
exten => s,n,Hangup()
[agent-logout]
exten => s,1,Answer()
exten => s,n,RemoveQueueMember(support)
exten => s,n,Playback(agent-loggedoff)
exten => s,n,Hangup()
[hotline]
exten => s,1,Answer()
exten => s,2,Set(TIMEOUT(response)=8)
rexten => 123,1,ChannelSpy(support,rq)
exten => s,3,Queue(support,tTwW)
exten => s,4,Background(/var/lib/asterisk/sounds/fr/recorded)
exten => s,5,WaitExten()
If these two lines are causally related, there is something rather strange in you definition of the user01 endpoint as you seem to have ended up calling your own extension 10, rather than the device. Is user01 a soft phone on the same machine, not that that would be sufficient for it to break in that way.
it’s my softphone
I’d suggest it may be claiming port 5060 on the registration, even though that port has already been taken by Asterisk. However, I don’t understand why the user part of the request URI would be 10.
I chose number 10 to connect to agents
infact he joins the queue of my trunk sip.
I use (pjsip tls) on my users and on my trunk sip (psip udp)
I feel stupid to understand nothing I only have problems it’s been a month since I would like to finalize my screenplay.
Since I joined the forum.
the motivation is indeed a moment the relentlessness no
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