[Asterisk-Users] VoicePulse Connect DTMF with IAX2

Michael Welter mike at introspect.com
Tue Aug 31 07:47:11 MST 2004


I'm using RC2 and last weekend's changes from VoicePulse.  Outbound 
calling and dtmf works fine.  However, an inbound call to my DID cannot 
send dtmf digits to the IVR.

Thoughts?



Deon Rodden wrote:
> 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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-- 
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
mike at introspect.com
www.introspect.com




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