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
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 <techsupport@progressivehomehealth.com> 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 > 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 >-- 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
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 <techsupport@progressivehomehealth.com> 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 >>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 >> >> >> > > >-- >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 >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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051117/cbd618d6/attachment.htm
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 <techsupport@progressivehomehealth.com> 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 > <techsupport@progressivehomehealth.com> 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 > 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 > > > > -- > 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 > 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 > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > >-- 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