We are using the same phones in our office with firmware 1.0.1.13 and have no issues.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Tuesday, February 14, 2006 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream hold one way audio -URGENT Yes, we have and we just got rid of them because of it. We use higher end phones like Polycom, Snom and Cisco now. On 2/14/06, Ronald Voermans <[EMAIL PROTECTED]> wrote: > > Hi all, > > At our customer site i've installed one asterisk server with 20 > Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the > customer, the receptionist picks up, and does an attended transfer > (the 'grandstream way') to a collegue. Most of the times this goes ok, > but sometimes, when the receptionist puts the call on hold, and tries > te reconnect to the caller there's only one-way-audio. The > receptionist can hear the caller, but the caller cannot hear the > receptionist! I've done several ngreps etc. and I can see that traffic > is going from asterisk to the receptionist phone, and vice versa. > > I can predict when this is going to happen: when the receptionist > places the call on hold, the caller doesn't hear musiconhold. If the > caller does hear musiconhold then everythings goes well. Asterisk > states in both occassions that it is starting musiconhold, and again, > with ngrep i can see the RTP traffic going from asterisk to the caller and vice-versa. > > I'm thinking this is a problem with the Grandstream phones, but I'm > not sure. I upgraded to of the phones to firmware 1.0.1.12 today, and > will contact the customer tomorrow if it had helped. Has anyone ever > seen this kind of behavior with Grandstreams/Asterisk? > > Thx, > > Ronald > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > 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 Fax: 518-631-2856 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users