# Message/ast\_msg\_queue stuck on Hangup() Application

**URL:** <https://community.asterisk.org/t/message-ast-msg-queue-stuck-on-hangup-application/77535>\
**Category:** Asterisk\
**Created:** [December 7, 2018, 5:00pm UTC](https://community.asterisk.org/t/message-ast-msg-queue-stuck-on-hangup-application/77535 "2018-12-07T17:00:32Z")\
**Posts on this page:** 1\
**Page:** 1

<div class="post-metadata">

**Author:** ![jcpierri](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcpierri/32/3435_2.png) [@jcpierri](https://community.asterisk.org/u/jcpierri)\
**Post date:** [December 7, 2018, 5:00pm UTC](https://community.asterisk.org/t/message-ast-msg-queue-stuck-on-hangup-application/77535/1 "2018-12-07T17:00:32Z")

</div>

I’m using asterisk 13.23.1 and I noted that any SIP message sent from one device to another, despite being correctly delivered to the intended destination, get stuck in the _Message/ast\_msg\_queue_ channel on the _Hangup()_ application, as shown here:

```auto
lab-asterisk-1*CLI> core show channels 
Channel Location State Application(Data)             
Message/ast_msg_queu 202@SIPmsgs:5 Up Hangup()                      
0 active channels
0 active calls
4 calls processed

```

My sip.conf is:

```auto
[general]
accept_outofcall_message=yes
outofcall_message_context=SIPmsgs

```

My extensions.conf is:

```auto
[SIPmsgs]
exten => _X.,1,Verbose(SIP message received)
 same => n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
 same => n,MessageSend(${ACTUALTO},${MESSAGE(from)})
 same => n,Verbose(Send status is ${MESSAGE_SEND_STATUS})
 same => n,Hangup()
 same => n,Hangup()

```

When a message is sent the Asterisk console shows:

```auto
SIP message received
Send status is SUCCESS

```

I’m using as reference [this document](https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_MessageSend)

Does anybody knows what I’m doing wrong to cause messages to stuck on Hangup()?

# EDIT

With SIP debug enabled, this is what appears when a message is sent from x202 to x201:

```auto
lab-asterisk-1*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:172.17.79.5:5070 --->
MESSAGE sip:201@172.17.77.234 SIP/2.0
Via: SIP/2.0/UDP 172.17.79.5:5070;branch=z9hG4bK152815559;rport
From: <sip:202@172.17.77.234>;tag=449170377
To: <sip:201@172.17.77.234>
Call-ID: 1093198086-5070-46@BHC.BH.HJ.F
CSeq: 250 MESSAGE
Contact: <sip:202@172.17.79.5:5070>
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2160 1.0.9.69
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: text/plain; charset=UTF-8
Content-Length: 12

teste novo 2
<------------->
--- (14 headers 1 lines) ---
Sending to 172.17.79.5:5070 (no NAT)
Receiving message!
Looking for 201 in SIPmsgs (domain 172.17.77.234)

<--- Transmitting (no NAT) to 172.17.79.5:5070 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 172.17.79.5:5070;branch=z9hG4bK152815559;received=172.17.79.5;rport=5070
From: <sip:202@172.17.77.234>;tag=449170377
To: <sip:201@172.17.77.234>;tag=as36efd2f6
Call-ID: 1093198086-5070-46@BHC.BH.HJ.F
CSeq: 250 MESSAGE
Server: Asterisk PBX 13.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '1093198086-5070-46@BHC.BH.HJ.F' in 32000 ms (Method: MESSAGE)
SIP message received
Reliably Transmitting (no NAT) to 172.17.79.21:5060:
MESSAGE sip:201@172.17.79.21:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.77.234:5060;branch=z9hG4bK512ce2ba
Max-Forwards: 70
From: "Unknown" <sip:202@172.17.77.234>;tag=as04e4b995
To: <sip:201@172.17.79.21:5060>
Contact: <sip:202@172.17.77.234:5060>
Call-ID: 7fcfce165486ad600022dd855e596426@172.17.77.234:5060
CSeq: 102 MESSAGE
User-Agent: Asterisk PBX 13.23.1
Content-Type: text/plain;charset=UTF-8
Content-Length: 12

teste novo 2
---
Scheduling destruction of SIP dialog '7fcfce165486ad600022dd855e596426@172.17.77.234:5060' in 6400 ms (Method: MESSAGE)
Send status is SUCCESS

<--- SIP read from UDP:172.17.79.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.77.234:5060;branch=z9hG4bK512ce2ba
From: "Unknown" <sip:202@172.17.77.234>;tag=as04e4b995
To: <sip:201@172.17.79.21:5060>;tag=3074602480
Call-ID: 7fcfce165486ad600022dd855e596426@172.17.77.234:5060
CSeq: 102 MESSAGE
User-Agent: Yealink SIP VP-T49G 51.80.0.120
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '7fcfce165486ad600022dd855e596426@172.17.77.234:5060' Method: MESSAGE
lab-asterisk-1*CLI> 

```

No mater how much time has elapsed, that channel remain stuck at Hangup().  
If Hangup() is removed, the channel get stuck on the immediately preceding command.