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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