Stuck channels after SIP calls

[mvega]

Hello All

I am wondering if anyone has some pointers as to why I would see a high number of channels hung after a call ends?

I am doing some load testing with SIPp on a set of test servers and found on one of the servers had high number of hung channels, hundreds. Not exactly sure what would cause this. The hung channels were causing high system resource usage (~96% CPU)

I’ve done some googling around but not seeing much in the way of root cause.

running Ast 14.6.2, channel request hangup all did shut the hung channels down. but dont know what caused them to hang in the first place.

Thanks for any input!

[jcolp]

If you can do “channel request hangup” then the channel isn’t hung. Instead nothing told it to hang up. This means either Asterisk never got a BYE or something else that would cause it to be hung up didn’t occur.

[mvega]

ah copy. thanks for the input

looking at the SIPp logs I am seeing a lot of SIP retransmission packets. that might explain that. If a call was started though, wouldn’t my extensions.conf hang it up after 20 seconds using this config however? regardless of if a BYE was sent from SIPp shouldn’t I expect Asterisk to end the call?

[sipp]
exten => 1001,1,Answer
exten => 1001,n,MusicOnHold()
#exten => 1001,n,MusicOnHold(,20)
exten => 1001,n,WaitMusicOnHold(20)
exten => 1001,n,Hangup

[rmudgett]

The MusicOnHold application will not exit unless you provide a value for the duration argument. Thus the dialplan you show will never initiate a hangup. Also the WaitMusicOnHold application does not exist.