SIP Staying Off-Hook, Help?!?

Asterisk Asterisk Support
1

I’m having a very real problem with SIP channels not hanging up at one particular site… they have the exact same hardware/config that we use everywhere, nobody is experiencing problems like them, and I can’t find what’s causing the problem.

This appears to be a station-side problem, but does not matter if the extension is local or remote. I’m getting open channels with all extensions.

Here’s what I’ve done:
Replaced entire chassis, hard drive, etc.
Originally 1.2.16 (I think), then did fresh build of 1.2.18
Replaced router
Moved from Cable to T1

P.S.
I’ve provided sample logs/conf below, and if you look specifically at the ‘sip show history’ xxxxxxxx, I think that’s where the problem lies (notice the 487 and continuing ACK)… I just don’t know what to do about it.

Any ideas?!?

  • J

asterisk*CLI> show channels 
Channel Location State Application(Data) 
0 active channels 
0 active calls

asterisk*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
192.168.1.21 210 5586207a065 00102/00000 unkn No Tx: ACK
192.168.1.21 210 37c9511e353 00102/00000 unkn No Tx: ACK
192.168.1.21 210 350a8e3e45e 00102/00000 unkn No Tx: ACK
67.188.24.233 230 31b027ba0ac 00102/00000 unkn No Tx: ACK
4 active SIP channels


asterisk*CLI> sip show history 37c9511e353 
asterisk*CLI> 
* SIP Call 
1. TxReqRel INVITE / 102 INVITE 
2. SchedDestroy 32000 ms 
3. Rx SIP/2.0 / 102 INVITE /100 Trying 
4. CancelDestroy 
5. TxReqRel CANCEL / 102 CANCEL 
6. SchedDestroy 32000 ms 
7. Rx SIP/2.0 / 102 INVITE /180 Ringing 
8. CancelDestroy 
9. Rx SIP/2.0 / 102 CANCEL /200 OK 
10. Rx SIP/2.0 / 102 INVITE /487 Request Terminated 
11. TxReq ACK / 102 ACK

/etc/asterisk/sip_nat.conf

nat=yes
externip=64.81.30.178
localnet=192.168.1.0/255.255.255.0
qualify=yes


/etc/asterisk/sip_additional.conf 

[210] 
type=friend 
secret=xxxxxxx 
record_out=Always 
record_in=Always 
qualify=no 
port=5060 
nat=no 
mailbox=210@default 
host=dynamic 
dtmfmode=rfc2833 
dial=SIP/210 
context=from-internal 
canreinvite=no 
callerid=device <210>

[230]
type=friend
secret=xxxxxxx
record_out=Always
record_in=Always
qualify=yes
port=5060
nat=yes
mailbox=230@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/230
context=from-internal
canreinvite=no
callerid=device <230>

2

This could(probably) be related to the SIP bug that caused channels not to be released. It affected versions from 1.2.x to 1.4.4. After upgrading it fixed this problem I had. Please check on bugs.digum.com, or just upgrade to latest 1.2.x, or 1.4.5

3

Yeah, I know about the bug, but I’m running the exact same setup here even & I haven’t got these problems… none of my sites do, just this one, and I don’t see what’s so special either. I’m going to upgrade it again to 1.2.20 prob in the next day or so (I saw 1.2.21, but we’ll see).

It may help to know that the inbound routing is nothing out of the ordinary either…

1.) Inbound call comes in with DID from Vitelity
2.) Call is pushed to an IVR where a greeting is heard
3.) The only option is transferring to a single queue
4.) A join message is played, CID is prefixed, and caller position is announced
5.) Caller speaks with agent (all calls recorded) or leaves message in voice mail (rare)

I’ve already tried:

1.) Changing proxies with Vitelity :frowning:
2.) Changing recording from per-agent to per-queue

They have a mix of Aastra 480iCT and Grandstream GXP-2000’s,

  • Jason
4

I sure would love to know what’s causing this. I’ve got two servers this issue happens on almost a daily basis. It’s a nuisance, especially since the stuck sip channels count against “restart when convenient”, as well as the count towards active calls on a sip peer, so if I have call-limit=1 set the phone becomes useless.

Sometimes, the stuck call eventually disappears, other times I have to restart asterisk to get rid of it.