[asterisk-dev] Urgent development consultancy wanted

[Alistair Cunningham]

We have a customer running Asterisk 1.8.7.1 who is suffering from stuck 
calls. The scenario is:

1. A call comes in from the PSTN via SIP.
2. We do a Dial() to a local channel.
3. In the local channel, we do a Dial() to a SIP URI which is a phone 
registered to OpenSIPS on a different machine.
4. The phone rings (and perhaps answers).
5. The caller hangs up.
6. Sometimes one of the channels (either the inbound channel or the 
local channel) never gets hung up, the "h" extension never gets called 
for it, and the channel remains in "core show channels" until Asterisk 
is restarted.

We're looking for a developer who is able to debug this urgently, 
preferably today. If anyone is available and has expertise at debugging 
this problem, please email me off-list with details of exactly when 
you're available, and of course your hourly rate.

-- 
Alistair Cunningham
+1 888 468 3111
+44 20 799 39 799
http://integrics.com/