[asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
Steve Davies
davies147 at gmail.com
Thu Mar 27 11:00:18 CDT 2008
On 27/03/2008, David Nedved <david_nedved at yahoo.com> wrote:
>
> So now it seems 1.4.18 is doing the same as 1.2.27 -- working for the
> most part but completely ignoring DTMF on incoming SIP calls.
>
Perhaps you now need to delve deeper. Capture a UDP trace between your
VoIP provider and Asterisk, and another of the same call between
Asterisk and a handset. Do this for an ordinary voice call, no IVR
menus etc etc.
1) Can you hear the DTMF being sent by the far end by the way?
2) If you use Wireshark to do a VoIP call analysis of the traces, do
you receive any DTMF signalling in the RTP stream, or in INFO packets
from your VoIP provider?
I'm sure there is more...
Steve
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