SIP Channels are not clearing periodically

Asterisk Asterisk SIP
1

Hi Group

I have a couple of Asterisk systems that SIP channels are locking up periodically. In both cases they are experiencing some fairly significant network issues which is likely causing the issue as below:

172.30.11.159 823 082cd26373c1424 (alaw) No Tx: ACK 823
172.30.11.159 (None) 0_1469382204@17 (nothing) No Rx: REGISTER
172.30.10.26 856 6e18a6394358906 (alaw) No Tx: ACK 856
172.30.13.136 (None) 0_4289754944@17 (nothing) No Rx: REGISTER
172.30.11.161 842 059b68b9028f968 (alaw) No Tx: ACK 842
172.30.11.130 821 0_3128238205@17 (alaw) No Tx: ACK 821
172.30.12.146 885 1170c15255fb2b0 (alaw) No Tx: ACK 885
172.30.11.131 847 5395b30b34d2255 (alaw) No Tx: ACK 847
172.30.11.2 (None) 1F9ECA17-4F5211 (alaw) No Rx: ACK gateway_is
172.30.11.159 823 0_2509359026@17 (alaw) No Tx: ACK 823
172.30.13.118 (None) 0_2458581150@17 (nothing) No Rx: REGISTER
172.30.11.125 880 0_1807668308@17 (alaw) No Tx: ACK 880
172.30.12.146 885 0_142646586@172 (alaw) No Tx: ACK 885
172.30.11.161 842 011f7d0e19ad070 (alaw) No Tx: ACK 842
172.30.11.125 880 0_3702533611@17 (alaw) No Tx: ACK 880

Most peers showing ACK are locked channels.

Excerpt from sip show inuse

  • Peer name In use Limit
    885 2/0/0 10
    880 2/0/0 10
    842 2/0/0 10
    823 2/0/0 10

sip reload does not clear the channels.

So my questions are:

  1. Is it a bug that this is occurring?
  2. If not, is there something that I can do to make sure these channels clear?
  3. Could this have something to do with RTCP?
  4. How can I clear the channels without restarting Asterisk?

Thanks so much all.

Regards
Michael Knill

2

Answered 4) channel request hangup

3

Unless SIP session timers or RTP timeout is configured and used then Asterisk relies on the remote side to state that the call is terminated. Depending on the call flow this can cause calls to remain up even if the remote side has gone away without Asterisk being told.

4

Thanks Josh for your help

Yes that will be it as I have disabled SIP session timers in sip.conf. I think this was done due to concerns of sessions being closed due to the timer.
As the problem seems only to be between the phone and Asterisk, Im interested in the best way to configure this e.g.:

  1. Set session-timers = accept in sip.conf and configure a session timer on the phones to say 1Hr?
  2. Set session-timers = originate in sip.conf for 1Hr and leave the phones session timer disabled (assuming they will accept a session timer)?
  3. Set session-timers = originate in sip.conf and on the phone for 1Hr?
  4. Something else?

Thanks
Mike

5

I haven’t dealt with session timers in enough detail to really answer that. You may want to experiment within your environment to see what works best for you.