Problem with extention status(busy)

hi everyone
randomly some extensions status change to busy ( in core show hints )
extension can receive call without any problem ,but blf keys shows extension busy when this problem was happend
when unregistered and then register the extension ,busy change too idle
please help me
sorry my language is terrible

What channel driver are you using?

sip channel
and last version of asterisk > 11.23.0

hi everyone , no help?

Are you configuring things in the .conf file or realtime? What is the output of “sip show channels”? What’s the console output? Are there any additional log messages?

what .conf file?
sip show channels don’t show me any channel with that extension
and there are not any log messages
what do u need for help?
i can get u any config file or log

How is chan_sip configured - from a sip.conf file, or have you put it into a database?
The console output is needed showing what has happened to the extension in question, and a general idea of how the system is used.

sip.conf file :

[general]
bindaddr=0.0.0.0
jbenable=yes
jbmaxsize=3
jbforce=yes
limitonpeer=yes
jbresyncthreshold=30
t38pt_udptl=yes
cisco_usecallmanager=yes
huntgroup_default=yes

how can i get console output?
problem not accourd by test somthing, happend randomly

Where are your peers and other things defined? There are none in what you have posted…

you told me send you sip.conf
peer defined to sip_additional.conf :

[2100]
deny=0.0.0.0/0.0.0.0
type=friend
sipserver=172.16.5.2
secret=2100
qualify=1000
port=5060
pickupgroup=21
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=2100@device
macaddr=38ED18559B2E
host=dynamic
dtmfmode=rfc2833
dial=SIP/2100
context=from-internal
clienttype=cisco
ciscoPhoneLabel=AFR@NET
ciscocallwaiting=3
canreinvite=no
callgroup=21
callerid=device <2100>
accountcode=
call-limit=2

cc_agent_policy=generic
cc_monitor_policy=generic
subscribecontext=ext-local
subscribe=2148
subscribe=2150
subscribe=2151
subscribe=2125
subscribe=8131
subscribe=8346



[2102]
deny=0.0.0.0/0.0.0.0
type=friend
sipserver=172.16.5.2
secret=2102
qualify=1000
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=2102@device
macaddr=38ED18559A69
host=dynamic
dtmfmode=rfc2833
dial=SIP/2102
context=from-internal
clienttype=cisco
ciscoPhoneLabel=AFR@NET
ciscocallwaiting=0
canreinvite=no
callgroup=
callerid=device <2102>
accountcode=
call-limit=2

cc_agent_policy=generic
cc_monitor_policy=generic
subscribecontext=ext-local
subscribe=2111
subscribe=2117
subscribe=2199
subscribe=2136

You are on an Asterisk forum, not a FreePBX forum. The breakdown by the use of #include is something done by FreePBX and Asterisk just processes the merged file.

1 Like

Yes, you are right saying

As well you appear to be using the massive Cisco patch to chan_sip. Does the issue occur without it?

maybe , i dont know issue accured for cisco path or not
what i need to know that?

what should i do? i need help

The Cisco patch makes quite a lot of changes to chan_sip, and it’s not something I’m personally familiar with or comfortable supporting a chan_sip build with. If you provide a log showing the device going to busy and not back when it shouldn’t someone may be able to help. I can also say no other issues come up in my mind that have been filed for this particular issue.

tnx for help
i try too get log for this issue and send for u