[Asterisk-Users] VoicePulse Connect DTMF with IAX2
Deon Rodden
drodden at webunited.net
Tue Aug 31 07:15:02 MST 2004
Here's my iax.conf and extensions.conf (I have not yet made the "recent"
changes they just emailed about a day ago, this is twice in a two month
period, jeesh) I have tested inbound and outbound dtmf. I use the g.711
codec and use inband.
iax.conf
--------------------------------------------------------------------------------------------------------------------------------------------------
[general]
port=5036
bindaddr=0.0.0.0
context=incoming
;iaxcompat=yes ; Set iaxcompat to yes if you plan to use
layered switches.
; It incurs a small performance hit to enable it.
delayreject=yes ; For increased security against brute force
password attacks.
; Enabling this will delay the sending of
authentication
; reject for REGREQ or AUTHREP if there is a
password.
amaflags=documentation ; global default AMA flag for iaxtel calls.
These flags
; are used in the generation of call detail records.
;accountcode=1 ; default account for Call Detail Records in
addition
; to specifying on a per-user basis.
language=en ; Global default language for users.
; If omitted, will fallback to english
bandwidth=high ; Specify bandwidth of low, medium, or high to
; control which codecs are used in general.
allow=all ; Which codecs to allow, same as bandwidth=high
disallow=g723.1 ; Hm... Proprietary, don't use it...
disallow=lpc10 ; Icky sound quality... Mr. Roboto.
; You can adjust several parameters relating to the jitter buffer.
; The jitter buffer's function is to compensate for varying network delay.
; All the jitter buffer settings except dropcount are in milliseconds.
; The jitter buffer works for INCOMING audio - the outbound audio
; will be dejittered by the jitter buffer at the other end.
;
jitterbuffer=no ; Whether you want the jitter buffer at all.
;dropcount=2 ; The jitter buffer is sized such that no more
than "dropcount"
; frames would have been "too late" over the
last 2 seconds.
; Set to a small number. "3" represents 1.5% of
frames dropped
;maxjitterbuffer=500 ; A maximum size for the jitter buffer. Setting
a reasonable maximum
; here will prevent the call delay from rising
to silly values in
; extreme situations.
;maxexcessbuffer=80 ; If conditions improve after a period of high
jitter, the jitter buffer
; can end up bigger than necessary. If it ends
up more than
; "maxexcessbuffer" bigger than needed, Asterisk
will start gradually
; decreasing the amount of jitter buffering.
;minexcessbuffer=80 ; Sets a desired mimimum amount of headroom in
the jitter buffer.
; If Asterisk has less headroom than this, then
it will start gradually
; increasing the amount of jitter buffering.
;jittershrinkrate=1 ; When the jitter buffer is being gradually
shrunk (or enlarged),
; how many millisecs shall we take off per 20ms
frame received?
; Use a small number, or you will be able to
hear it changing.
; An example: if you set this to 2, then the
jitter buffer size will
; change by 100 millisec per second.
;trunkfreq=20 ; How frequently to send trunk msgs (in ms)
authdebug=no ; You can disable authentication debugging to reduce
; the amount of debugging traffic.
tos=lowdelay ; You can set values for your TOS bits to help
improve performance.
; Can be lowdelay, throughput, reliability,
mincost or none.
;mailboxdetail=yes ; If set to "yes", the user receives the actual
; new/old message counts, not just a yes/no as to
; whether they have messages.
register => in-xxx##XxX#X:xXx##Xxx##@gw5.voicepulse.com
; ### PROVIDERS ###
[voicepulse] ; For inbound
context=VPWS
type=user
host=gw5.voicepulse.com
accountcode=1
[vpconnect-t01] ; For outbound
type=peer
secret=xXx##Xxx##
host=gwiaxt01.voicepulse.com
auth=md5
qualify=yes
accountcode=1
[vpconnect-t02] ; Outbound backup
type=peer
secret=xXx##Xxx##
host=gwiaxt02.voicepulse.com
auth=md5
qualify=yes
accountcode=1
--------------------------------------------------------------------------------------------------------------------------------------------------
extensions.conf
--------------------------------------------------------------------------------------------------------------------------------------------------
[VPWS]
; All Inbound Voicepulse DID numbers go here
; From here it is distributed to the propper place
;; - Some Company -
exten => 1235551212,1,Goto(company,1235551212,1)
[company]
; Local
exten => _NXXXXXX,1,Dial(IAX2/xxx##XxX#X at vpconnect-t01/1304${EXTEN})
exten => _NXXXXXX,2,Dial(IAX2/xxx##XxX#X at vpconnect-t02/1304${EXTEN})
exten => _NXXXXXX,3,Congestion
exten => _NXXNXXXXXX,1,Dial(IAX2/xxx##XxX#X at vpconnect-t01/1${EXTEN})
exten => _NXXNXXXXXX,2,Dial(IAX2/xxx##XxX#X at vpconnect-t02/1${EXTEN})
exten => _NXXNXXXXXX,3,Congestion
; Long Distance
exten => _1NXXNXXXXXX,1,Dial(IAX2/xxx##XxX#X at vpconnect-t01/${EXTEN})
exten => _1NXXNXXXXXX,2,Dial(IAX2/xxx##XxX#X at vpconnect-t02/${EXTEN})
exten => _1NXXNXXXXXX,3,Congestion
exten => 1235551212,1,Dial(SIP/whoever)
--------------------------------------------------------------------------------------------------------------------------------------------------
Brian Capouch wrote:
> Bryce Nesbitt (mailing list account) wrote:
>
>> Is there anyone out there who has VoicePulse Connect working with DTMF?
>> I've been unable to get it to work from the start, and the recent
>> VoicePulse updates
>> did not help.
>>
>>
>
> I use VoicePulse connect, have similar configs (although I only use
> iLBC with them) and things are working just fine for me. I just
> tested with CVS from a day or two ago. I call out and can do DTMF
> stuff, and likewise if I call in to my DID the caller can navigate my
> IVRs just fine with DTMF.
>
> A data point, I guess. Are you using recent CVS?
>
> B.
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