Billy Huddleston wrote: > change dtmf to info on both * and in the handytone. > > ----- Original Message ----- > From: "Senad Jordanovic" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > 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 >> >> _______________________________________________ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users
My understanding from this months GS related posts is that "info" is not sending the digits properly. Is that the case with you? _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
