[asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP

[David Nedved]

--- "Darrick Hartman (lists)" <dhartman at djhsolutions.com> wrote:
> Do yourself a favor and upgrade a Asterisk 1.4 which has a proper 
> implementation of DTMF.  It's likely your SIP provider upgraded to 
> something which does not recognize the DTMF tones from Asterisk 1.2.

I've upgraded to 1.4.18 (along with zaptel 1.4.9.2) and still
experiencing the same problem (not recognizing DTMF on SIP inbound
calls) as well as new problems.  The new problems are much more severe
than the previous problems so I'm starting a new thread with a more
descriptive subject.  I've changed sip.conf to eliminate warnings for
new syntax:

insecure=port,invite
dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info

Everything else is as-was in sip.conf, extensions.conf, iax.conf,
rtp.conf, voicemail.conf, zapata.conf, zaptel.conf (although I looked
through the new samples and didn't see anything glaring I needed to
change).  For the config files I had not changed I took the new sample
files.

Now in addition to not recognizing DTMF on SIP still, asterisk is now
frequently dropping calls when I start to enter DTMF.  On console I get
lines such as:

    -- Executing [xxxxx at incoming-viatalk:1] Goto("SIP/xxxxx-081ea720",
"incoming|s|1") in new stack
    -- Goto (incoming,s,1)
    -- Executing [s at incoming:1] Answer("SIP/xxxxx-081ea720", "") in new
stack
    -- Executing [s at incoming:2] BackGround("SIP/xxxxx-081ea720",
"/home/dnedved/hello") in new stack
    -- <SIP/xxxxx-081ea720> Playing '/home/dnedved/hello' (language
'en')
  == Auto fallthrough, channel 'SIP/xxxxx-081ea720' status is 'UNKNOWN'

It's also happening on zaptel channels (although not nearly so
frequently), so it's not a SIP only problem:

[Mar 27 10:42:07] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 18
(Ring Begin)...
[Mar 27 10:42:08] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 2
(Ring/Answered)...
[Mar 27 10:42:12] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 18
(Ring Begin)...
    -- Executing [s at line_nl:1] Goto("Zap/4-1", "incoming|s|1") in new
stack
    -- Goto (incoming,s,1)
    -- Executing [s at incoming:1] Answer("Zap/4-1", "") in new stack
    -- Executing [s at incoming:2] BackGround("Zap/4-1",
"/home/dnedved/hello") in new stack
    -- <Zap/4-1> Playing '/home/dnedved/hello' (language 'en')
  == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'
    -- Hungup 'Zap/4-1'

I don't know much about asterisk debugging since it has worked so
flawlessly so far, but I would guess that the Auto fallthrough with
status UNKNOWN means that the application that was running died, didn't
set any return code, so asterisk dropped the call?  I'm running in
console mode with 5 v's of verbose mode, how do I find more information
about why it's dropping these calls?

Thanks and best regards,

David

david_nedved at yahoo.com


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