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

[Steve Totaro]

On Thu, Mar 27, 2008 at 8:16 AM, Darrick Hartman (lists)
<dhartman at djhsolutions.com> wrote:
> David Nedved wrote:
>  > --- "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.
>
>  There were several things that changed...
>
>
>  > 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'
>
>  Try adding this line in the general section of extensions.conf
>
>  autofallthrough=no
>
>  The default behavior in 1.2 was no.  In 1.4 it changed to yes.  That
>  will be your simplest fix (without seeing your dialplan).  Asterisk is
>  moving on to the next step in the dialplan before you enter your digits.
>   You need to have it wait for the digits to be entered.
>
>  Darrick
>  --
>  Darrick Hartman
>  DJH Solutions, LLC
>  http://www.djhsolutions.com
>  http://www.djhsolutions.com/wiki
>


If the DTMF issue works better in 1.2.X and you do not need the
additional features of 1.4.X then you made the right choice going back
to 1.2.X.

People on the list (mainly dev) want you to test, find bugs, jump
through hoops, and post to Mantis (where you bug might just be closed,
or a general feeling of "You are wrong".  All of this testing is free
of course due to the "Benefit of the Community".

In the real world, it would serve you better to do what works best for
your business.  Don't let the "Dev" guys push you around, do what
makes sense to your business.

Thanks,
Steve Totaro