[asterisk-users] E&M wink/no audio

[Bill Michaelson]

I am preparing to connect an asterisk box with a redfone fonebridge to a 
T1 service provider.  I am doing this by testing first with another 
asterisk and a Sangoma card playing the role of telco.

I formerly had this test configuration operating flawlessly as a PRI 
connection.  But I discovered that I will need to use E&M, thus I've 
chosen the parameters as described in the subject line.

So far, I am able to initiate a call from the Sangoma/telco side to the 
fonebridge side, and basic robbed bit/ABCD call supervision seems to 
work.  That is, I can see the flags going up and down on but sides with 
zttool, and a Zap channel is allocated on each side.  But it seems that 
DNIS is not going through, and so I had to fudge it by creating a "s" 
extension on the called side to pass the call through to a SIP 
telephone.  When the call is answered, the caller hears silence, and the 
call recipient hears a soft squeal.

Am I being reasonable in assuming that the absence of a valid audio 
stream is the likely reason that the called number is not being passed 
through successfully?  In any case, what type of stuff should I be 
looking for to diagnose this?

FYI, the calling side issues a log message that seems relevant, but it's 
precise implications elude me: chan_zap.c: Ignoring wink on channel 1 
(see below).  I hope someone can give clue on these matters.

[Sep 22 11:27:05] VERBOSE[27487] logger.c: -- Executing 
[s at macro-dialout-trunk:19] Dial("SIP/366-a41a4560", 
"ZAP/g1/12223336666|300|wW") in new stack
[Sep 22 11:27:05] DEBUG[27487] chan_zap.c: Dialing '12223336666'
[Sep 22 11:27:05] DEBUG[27487] chan_zap.c: Deferring dialing...
[Sep 22 11:27:05] VERBOSE[27487] logger.c: -- Called g1/12223336666
[Sep 22 11:27:06] DEBUG[27487] chan_zap.c: Ignoring wink on channel 1
[Sep 22 11:27:06] DEBUG[27487] chan_zap.c: Sent deferred digit string: 
T12223336666w
[Sep 22 11:27:19] VERBOSE[27487] logger.c: -- Hungup 'Zap/1-1'

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