Call gets disconnected after 30 seconds

Asterisk Asterisk Support
1

Hello,

I have installed asterisk-1.6.0.26, asterisk-addons-1.6.0.4, dahdi-linux-2.2.1.1, dahdi-tools-2.2.1.1.
When I dial a toll free number from pstn line to SIP sotf phone, the call is connected and is working fine.there is no problem.

I am having SIP agents,and they make login to asterisk. and when any incoming call comes, it is landing to the agent.
I dial the toll free number and try to land the call to my SIP agents through queue, the incoming call lands to the SIP agent. there is no probem. But the call gets disconnected after 25 to 30 seconds. Even the SIP agents get logged out.

My extensions.conf

[from-pstn]
exten => _X.,1,Answer()
exten => _X.,n,Queue(Process-1-English,tT,300)

My agents.conf

agent => 1000
agent => 2000

My Queues.conf

[Process-1-English]
strategy = ringall
context = Process-1-English-callback
timeout = 15
wrapuptime=5 ;15
announce-frequency = 30
announce-holdtime = yes
announce-position = yes
joinempty = yes
member => Agent/1000
member => Agent/2000

==============
My Error Log :

*CLI>
<— SIP read from UDP://192.168.1.36:52266 —>

BYE sip:888@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.36:52266;branch=z9hG4bK-d8754z-4864e215d21de56e-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2000@192.168.1.36:52266
To: "888"sip:888@192.168.1.2:5060;tag=as2532a6c5
From: "2000"sip:2000@192.168.1.2:5060;tag=a31b335b
Call-ID: N2JlZWU0NjFhMTAxZDM5MTA2MzZjYTRkMDU4YzQ5MjM.

CSeq: 5 BYE
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username=“2000”,realm=“asterisk”,nonce=“00fc1bd6”,uri="sip:888@192.168.1.2",response=“1e6ca8963469714cc57df9a6b11b95a3”,algorithm=MD5

Reason: SIP;description="User Hung Up"
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.36 : 52266 (no NAT)

<— Transmitting (no NAT) to 192.168.1.36:52266 —>

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.36:52266;branch=z9hG4bK-d8754z-4864e215d21de56e-1—d8754z-;received=192.168.1.36;rport=52266
From: "2000"sip:2000@192.168.1.2:5060;tag=a31b335b
To: "888"sip:888@192.168.1.2:5060;tag=as2532a6c5
Call-ID: N2JlZWU0NjFhMTAxZDM5MTA2MzZjYTRkMDU4YzQ5MjM.

CSeq: 5 BYE
User-Agent: Asterisk PBX 1.6.0.26
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
[Jun 7 12:13:52] ERROR[6886]: res_config_mysql.c:868 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on 192.168.1.35 (err 2003). Check debug for more info.
== Spawn extension (Process-1-English, _X., 5) exited non-zero on ‘DAHDI/1-1’
– Hungup ‘DAHDI/1-1’
[Jun 7 12:13:55] ERROR[6884]: res_config_mysql.c:868 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on 192.168.1.35 (err 2003). Check debug for more info.
== Agent ‘2000’ logged out
== Spawn extension (default, 888, 1) exited non-zero on 'SIP/2000-00000001’
Really destroying SIP dialog ‘N2JlZWU0NjFhMTAxZDM5MTA2MzZjYTRkMDU4YzQ5MjM.’ Method: BYE

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Thanks for your support.

-Urmi

2

X-Lite’s will drop out in that sort of time frame if you try to enable re-invites. Your SIP trace is incomplete, but I’m not convinced that is the problem. The trace you include shows a normal clearing by the X-Lite. Lack of RTP might cause the X-Lite to drop the call, but there isn’t enough relevant trace.

You do seem to have a database problem, but that is probably unrelated.

3

Hello,

To fix the issue you need to remove the check mark from the advance option configuration on the option settings. I had the same issue. Go to Options then advance on the bottom left cornet and at the middle of the configuration remove the check mark.

I hope this will resolve your problem.

Kind regards,

Al