[Asterisk-Users] Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9

[Shad Mortazavi]

Dear All,

I have been running an Asterisk 0.7.1 (patched with various agent
applications) server for almost 2 years. 

We have a data center in the USA and a call center in the UK. All calls
are routed to a group of central call queues in the USA. Agents from the
data center, call center and from remote locations (London, Scotland,
LA, Florida, and Maine) can log in, join the call queue and pick up
calls

This function has worked well since implementing the system and works
well using SNOM 200's (data center and call center) and SJ Phone Build
1.50.271d, Mar 11 2005.

I have rebuilt an identical test environment in my test lab and I can
run version 0.7.1 (patched). I log in as an agent using my softphone,
make a call from a second phone, I get greeted, put in a queue, given my
position, the call goes through to my soft phone, I accept the call,
press # and I'm on the call.

I run the upgrade to version 1.0.9 and run the same test;

I get greeted, put in a queue, given my position, the call goes through
to my soft phone, and I accept the call, press #... I then get a message
telling me that the system saying transfer? I see nothing on the CLI
except the usual waiting for '#' to acknowledge

To discount the SJ Phone I installed the version of X-Ten light that
some of our agents/staff use and I got the same result. I checked the
DTMF setting in sio.conf and these appear correct.

I downgrade to 0.7.1 and the function works on both SJ Phone and X-ten
light.

I have included the CLI captures below;

Sip show agents;

(Angela Holt) available at '0401 at sip' (musiconhold is 'default')

>From the CLI>

-- outgoing agentcall, to agent '1031', on 'Local/0401 at sip-bcc2,1'
    -- Called Agent/1031
    -- Executing Dial("Local/0401 at sip-bcc2,2",
"SIP/phone6&SIP/0401|20|tr") in new stack
    -- Called phone6
    -- Called 0401
    -- Agent/1031 is ringing
    -- SIP/phone6-1d2b is ringing
    -- Agent/1031 is ringing
    -- SIP/phone6-1d2b answered Local/0401 at sip-bcc2,2
    -- Local/0401 at sip-bcc2,1 answered, waiting for '#' to acknowledge

I need to move to the latest version of Asterisk to enable me to measure
the number of minutes a user has been held in a queue. This function was
not available in version 0.7.1. I remember a similar problem with
version 0.7.2.

Anyone else run into the same issue? 
Is it a known issue/bug? 
What is the fix?

Thanks and Regards

Shad Mortazavi
------------------------------------------------------
Nexus Global Technical Manager
n|m Nexus Management Inc