[Asterisk-Users] SIP calls no longer hangup [1.0.8]

[Ronan Mullally]

Hi,

I've just upgraded by asterisk box from 1.0.7 to 1.0.8 / 1.0.9. 
I'm running Gentoo, and in the UK, on a BT PSTN line.

The box has been running more or less fine for several months.
Since upgrading asterisk has been failing to hangup inbound /
outbound calls.  I've kept my original config files.

The sequence of events is roughly:

  - Place call from a cisco 7940 through the asterisk box to an
    outbound PSTN line
  - Far end answers (or not)
  - I decide to hangup the SIP phone (regardless of answer status)
  - asterisk doesn't see this hangup, but does log
     "chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call"
  - asterisk keeps the PSTN line up until I do a 'soft hangup'
    on the SIP channel.

  - The same thing appears to happen for SIP-SIP calls.

  - Outbound calls made from a ZAP FXS channel work fine.

  - For inbound calls, the call is set up fine, but regardless of
    which end terminates the call both the ZAP and SIP channels remain
    up.

  - If the inbound call is answered by a ZAP phone then the call is
    cleared fine.

  - I've also noticed that when placing or receiving calls from the SIP
    phone that no audio appears to be transmitted from the SIP phone.
    Again, this works fine from a ZAP handset.

This has happened on every call I've made since upgrading this afternoon.
I've reverted back to 1.0.7 (almost - the closest match now in Gentoo's
portage tree is 1.0.7-r1) but it still doesn't work.

I've had a look on voip-info and googled around a bit, but can't find 
anything.  Has anybody seen this before?  Have there been any code
changes in the past few months that might cause this?


-Ronan