[Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # to work during a call

2005-06-16 Thread Michael J. Tubby B.Sc (Hons) G8TIC



Gents,
 
I've built an Asterisk system to replace our PBX at 
work and have Cisco
7960 phones (SIP 7.4) running with Asterisk 
1.0.7.
 
How to I get Asterisk to recognise the '#' being 
pressed during a call?
 
In sip.conf I have entries likle this:
 
    [2001]    
type=friend    context=local-phone    
auth=md5    username=2001    
secret=xyzzy    callerid=Jack Tubby 
<2001>    host=dynamic    
nat=no    canreinvite=no    
dtmfmode=rfc2833    incominglimit=2    [EMAIL PROTECTED]    
disallow=all    allow=alaw    
allow=ulaw    callgroup=2    
pickupgroup=2
and in the SIPDefault.cnf for the phones I 
have:
 
    # Inband DTMF Settings 
(0-disable, 1-enable (default))    dtmf_inband: 
1
 
    # Out of band DTMF Settings 
(none-disable, avt-avt enable (default), avt_always - always avt 
)    dtmf_outofband: avt
 
    # DTMF dB Level Settings (1-6dB 
down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)    
dtmf_db_level: 3
DTMF works for voicemail and for remote services 
over both analogue Zap
channels and digital (ISDN) channels.
 
Asterisk doesn't appear to be 'monitoring' the 
audio so I can't get to Asterisk
features like Asterisk's transfer, parked calls and 
one-tuch-record...
 
Am I missing something?
 
 
Mike
 
 
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Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # to work during a call

2005-06-16 Thread Andrew Latham
# and * are mapped later in the SIP(Default/MAC).cnf it has a section
in the manual if you want to see why.

On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC <[EMAIL PROTECTED]> wrote:
>  
> Gents, 
>   
> I've built an Asterisk system to replace our PBX at work and have Cisco 
> 7960 phones (SIP 7.4) running with Asterisk 1.0.7. 
>   
> How to I get Asterisk to recognise the '#' being pressed during a call? 
>   
> In sip.conf I have entries likle this: 
>   
> [2001]
> type=friend
> context=local-phone
> auth=md5
> username=2001
> secret=xyzzy
> callerid=Jack Tubby <2001>
> host=dynamic
> nat=no
> canreinvite=no
> dtmfmode=rfc2833
> incominglimit=2
> [EMAIL PROTECTED]
> disallow=all
> allow=alaw
> allow=ulaw
> callgroup=2
> pickupgroup=2
>  
> and in the SIPDefault.cnf for the phones I have: 
>   
> # Inband DTMF Settings (0-disable, 1-enable (default))
> dtmf_inband: 1 
>   
> # Out of band DTMF Settings (none-disable, avt-avt enable (default),
> avt_always - always avt )
> dtmf_outofband: avt 
>   
> # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default),
> 4-3db up, 5-6dB up)
> dtmf_db_level: 3
>  
> DTMF works for voicemail and for remote services over both analogue Zap 
> channels and digital (ISDN) channels. 
>   
> Asterisk doesn't appear to be 'monitoring' the audio so I can't get to
> Asterisk 
> features like Asterisk's transfer, parked calls and one-tuch-record... 
>   
> Am I missing something? 
>   
>   
> Mike 
>   
>   
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 


-- 

Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!

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