Update:
I was able to obtain another VSP to try and rule out Broadvoice. Seems
that either my Broadvoice settings, or something on their end is causing
the brief screech noise upon playing the first sound.
However, with this new VSP I still have the AMD (Answering Machine
Detect) problem where it locks up unless I play some sound before
calling AMD. So my modified question is, has anyone ever had a problem
with AMD through a VSP (SIP, in this case). And it does *not* lock up
when calling phones local to the server.
Christopher Robinson wrote:
Bear with me this is a bit long winded. I am having some issues
making automated outbound calls over Broadvoice from my Asterisk 1.4.2
server. For reference, none of the below issues happen when I make
the calls to VoIP phones attached to the Asterisk server. What I am
trying to do is call, using a .call file, out via the SIP trunk we
have setup, and when the party picks up use AMD to detect if it's
reached a human or machine. If it's human then one message will be
played, and if machine another will be played theoretically after the
answering machine/voicemail is done playing. By the way, I'd like to
mention that this is not at all for spamming, or telemarketing. This
is an appointment reminder service.
from extensions.conf:
[mycontext]
exten => 899,1,Answer
exten => 899,2,Wait(2)
exten => 899,3,AMD
exten => 899,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
exten => 899,n(mach),WaitForSilence(2500)
exten => 899,n,Playback(were-sorry)
exten => 899,n,Hangup
exten => 899,n(humn),WaitForSilence(500)
exten => 899,n,Playback(welcome)
exten => 899,n,Hangup
The call goes out fine. When I pick it up AMD basically locks up,
although not exactly because as you can see below it does recognize
the HANGUP. However, it will not recognize my voice or dead air no
matter how long I stay on the call to try. If I just let my voicemail
pickup it does the same thing...takes forever for the call to
terminate. Again, this all works as expected when I make the call to
a SIP phone attached to the Asterisk server.
-- Attempting call on SIP/[EMAIL PROTECTED] for
[EMAIL PROTECTED]:1 (Retry 1)
> Channel SIP/sip.broadvoice.com-08bad080 was answered.
-- Executing [EMAIL PROTECTED]:1]
Answer("SIP/sip.broadvoice.com-08bad080", "") in new stack
-- Executing [EMAIL PROTECTED]:2]
AMD("SIP/sip.broadvoice.com-08bad080", "") in new stack
-- AMD: SIP/sip.broadvoice.com-08bad080 (null) (Fmt: 4)
-- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence
[800] totalAnalysisTime [5000] minimumWordLength [100]
betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256]
-- AMD: HANGUP
I did find a solution to this "lock up". That was to play a bit of
silence at any point before I actually call AMD (even before Answer
works):
[mycontext]
exten => 899,1,Playback(silence/1)
exten => 899,2,Answer
....
Although I don't particularly like this solution, as I'm just patching
the problem that I still don't understand, plus it adds a little more
delay that confuses the called party.
Also, when I tried this I realized yet another issue, which could be
the underlying cause of the whole thing. No matter what sound it is,
no matter if I use AMD or not, the very first sound that I play
results in a short "screech" sound before it is played. This happens
every time without fail. If I were to guess, I would say that there
is some data in the audio channel that is not audio data, and is being
represented with that screech sound...but of course that's just a guess.
Any help would be greatly appreciated. Below are some relevant
configuration settings:
sip.conf:
[general]
context=testusers ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support.
(Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard
port is 5060)
externip=xx.xx.xx.xx
localnet=192.168.1.0/255.255.255.0
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
pedantic=no
register =>
[EMAIL PROTECTED]:mysecret:[EMAIL PROTECTED]
[sip.broadvoice.com]
allow=ulaw
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=716XXXXXXX
secret=mysecret
username=716XXXXXXX
insecure=very
context=from_broadvoice
authname=716XXXXXXX
dtmf=inband
dtmfmode=inband
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=yes
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