[asterisk-users] e&m wink, TE110P, * answers too soon

[Steve Linabery]

Hi,

I've been googling all over the place and have read the relevant articles in the Digium knowledge base. I have tried all the suggestions I found in the K.B. Spent some time on the asterisk irc, tweaking some parameters as people thereon thought would be helpful, but to no avail.

I am trying to set up * on an e&m wink trunk currently attached to an Avaya Merlin Magix system. The provider of the T1 is McLeodUSA; our location is St Paul MN USA. I am in the process of getting more specific timing information from their tech support, but it takes days.

I can call into the * PBX from my cell phone just fine. I can call between the two grandstream phones I bought for testing just fine.

Here's the problem. When a call comes into *, * attempts to route it to an extension prematurely. For example, if the DTMF digits coming from upstream are '538', * tries to send the call to extn '53'. I still receive the '8', but too late.

Here's a snip from /var/log/asterisk/messages where the incoming DID digits are '535':
Aug  7 22:30:00 DEBUG[31492] chan_zap.c: Monitor doohicky got event Ring/Answered on channel 1
Aug  7 22:30:00 DEBUG[31478] devicestate.c: Changing state for Zap/1 - state 2 (In use)
Aug  7 22:30:00 VERBOSE[31493] logger.c: Asterisk Ready.
    -- Starting simple switch on 'Zap/1-1'
Aug  7 22:30:00 DEBUG[31494] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Aug  7 22:30:01 DEBUG[31493] chan_zap.c: DTMF digit: 5 on Zap/1-1
Aug  7 22:30:01 DEBUG[31493] chan_zap.c: DTMF digit: 3 on Zap/1-1
Aug  7 22:30:01 DEBUG[31493] chan_zap.c: Enabled echo cancellation on channel 1
Aug  7 22:30:01 VERBOSE[31493] logger.c:   == Unknown extension '53' in context 'demo' requested
Aug  7 22:30:04 DEBUG[31493] channel.c: Set channel Zap/1-1 to write format gsm
Aug  7 22:30:04 DEBUG[31493] channel.c: Scheduling timer at 160 sample intervals
Aug  7 22:30:04 VERBOSE[31493] logger.c:     -- Playing 'ss-noservice' (language 'en')
Aug  7 22:30:04 DEBUG[31493] chan_zap.c: DTMF digit: 5 on Zap/1-1
Aug  7 22:30:07 DEBUG[31493] chan_zap.c: Exception on 20, channel 1
Aug  7 22:30:07 DEBUG[31493] chan_zap.c: Got event On hook(1) on channel 1 (index 0)
Aug  7 22:30:07 DEBUG[31493] chan_zap.c: disabled echo cancellation on channel 1
Aug  7 22:30:07 DEBUG[31493] channel.c: Scheduling timer at 0 sample intervals
Aug  7 22:30:07 DEBUG[31493] channel.c: Hanging up channel 'Zap/1-1'
Aug  7 22:30:07 DEBUG[31493] chan_zap.c: zt_hangup(Zap/1-1)
Aug  7 22:30:07 DEBUG[31493] chan_zap.c: Hangup: channel: 1 index = 0, normal = 20, callwait = -1, thirdcall = -1
Aug  7 22:30:07 DEBUG[31493] chan_zap.c: disabled echo cancellation on channel 1
Aug  7 22:30:07 DEBUG[31493] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
Aug  7 22:30:07 DEBUG[31493] chan_zap.c: Updated conferencing on 1, with 0 conference users
Aug  7 22:30:07 VERBOSE[31493] logger.c:     -- Hungup 'Zap/1-1'
Aug  7 22:30:07 DEBUG[31478] devicestate.c: Changing state for Zap/1 - state 0 (Unknown)
Aug  7 22:30:07 DEBUG[31495] app_queue.c: Device 'Zap/1' changed to state '0' (Unknown) but we don't care because they're not a member of any queue.


Here are some settings from /etc/asterisk/zapata.conf:
[trunkgroups]
[channels]
wink=300
rxwink=300
start=3000
context=default
switchtype=national
toneduration=100
usecallerid=no
cidsignalling=dtmf
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=no
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
callprogress=no
switchtype = national
context = demo
signalling = em_w
group = 1
channel => 1-20


It has occurred to me that I could just set immediate=yes, read the incoming DTMF digits into a variable, and route to the appropriate extension. That seems more fragile to me since we could someday (when I'm not here) start getting more than 3 digits (caller id, for example). Plus I'd like to make it work the way it's *supposed* to.

Any help/suggestions are appreciated!

Cheers,
-- 
Steve Linabery
B94B C3C7 8A27 FF09 3C9D  E992 5A20 2492 D5F5 EE51


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