Ronald Voermans
2006-Feb-14 13:05 UTC
[Asterisk-Users] Grandstream hold one way audio -URGENT
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060214/83cbd00f/attachment.htm
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 <r.voermans@global-e.nl> 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
asterisk@anime.net
2006-Feb-14 14:49 UTC
[Asterisk-Users] Grandstream hold one way audio -URGENT
On Tue, 14 Feb 2006, Ronald Voermans wrote:> 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.if you have qualify=yes on the grandstream extensions, try taking it out. -Dan
We are using the same phones in our office with firmware 1.0.1.13 and have no issues. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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 <r.voermans@global-e.nl> 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 callerand 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