It would not have affected you in 1.0.1.9 if you set the DTMF mode to another setting. Mine all got reset when I upgraded to 1.0.12 and 3 of the phones I used had to be factory reset and then apply the .12 for them to work properly.
On 11/17/05, Health Masters <[EMAIL PROTECTED]> wrote: > We will check that... but that would have affected us in 1.0.1.9 correct? > Im inclined to believe this is a phone problem less an * prob. I dont > understand the changing from pcmu to gsm while on hold > can someone explain if it is supposed to work like this. > > Does the community have any influence with Grandstream? I have been > watching the wiki seems we are logging allot of issues and wish list. > > > Tom Vile wrote: > Check your DTMF setting on the phone and make sure it matches your > extension in Asterisk, the default in the GXP-2000 is INBAND and you > may have it set differently in Asterisk. > > On 11/16/05, Health Masters > <[EMAIL PROTECTED]> wrote: > > > We currently use the Grandstream GXP-2000 for our sip phones. Today we > upgraded our firmware to 1.01.12, it fixed allot of echo issues > especially on speaker. We found that after the upgraded that internal > users that put other internal users on hold are unable to regain the > call. In order to repick up the call the initiating user has to pick up > the far user off of hold and be placed on hold by the far user. We have > tested many times and this is the only way to regain the call. I dont > know if it matters but when the original user places the far user on > hold the music does not play and will only play when the second instance > of hold is done. Very strange any Ideas ? Any calls originating outside > of the sip network can be placed on hold with out problem,would this > have anything to do with zap versus sip? Also on the display when first > put on hold you can see that the call changes from pcmu to gsm then when > taken off hold goes to G.### until the other person has put you on hold > and picked you up does it go back to pcmu. > > Thanks > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Tom Vile > Baldwin Technology Solutions, Inc > Consulting - Web Design - VoIP Telephony > www.baldwintechsolutions.com > Phone: 518-631-2855 x205 > Phone: 978-203-3848 x205 > Fax: 518-631-2856 > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
