[Asterisk-Users] Handytone 286 - calling out

[Senad Jordanovic]

Billy Huddleston wrote:
> change dtmf to info on both * and in the handytone.
> 
> ----- Original Message -----
> From: "Senad Jordanovic" <senad at boltblue.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Tuesday, November 25, 2003 8:01 PM
> Subject: [Asterisk-Users] Handytone 286 - calling out
> 
> 
>> Hi,
>> 
>> Just received recently released Grandstream handytone 286 ATA for
>> testing. 
>> 
>> I can call ATA from any other extensions and conversations seems to
>> be of quite good quality. However placing calls from ATA is not
>> possible at all to any extensions. After dialing there no
>> indications of any kind from ATA at all. It just "hangs in there".
>> 
>> ATA is behind NAT, registers to an * with public IP with no problems
>> and it uses 1.0.4.17 firmware. Web config screen has detected
>> "firewall/NAT type is open Internet" as network firewall.
>> 
>> Here is my sip.conf:
>> [2202]
>> callerid="HandyTone" <2202>
>> username=2202
>> context=intern
>> qualify=500
>> type=friend
>> secret=XXXXXX
>> host=dynamic
>> dtmfmode=inband
>> canreinvite=no
>> reinvite=no
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> 
>> Any suggestions/pointers will be appreciated.
>> 
>> Ta
>> SJ
>> 
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>> 
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My understanding from this months GS related posts is that "info" is not
sending the digits properly.
Is that the case with you?