[asterisk-dev] Urgent development consultancy wanted

[Kevin P. Fleming]

On 05/22/2012 06:44 AM, Alistair Cunningham wrote:
> 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.

It is possible that the patch in revision 365896 could be related, 
although the symptoms sound slightly different.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org