[Asterisk-Users] Handytone 286 - calling out

[Billy Huddleston]

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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