Hi Noah, 1 - my asterisk version is 1.2.18 2 - my SIP devices are SNOM phones 3 - no SIP provider is involved...they are connected to my Asterisk...this is the strangest thing. This happens sometimes....I think it could be a network overload...can it be?
TIA Giorgio Noah Miller wrote: > Hi Giorgio - > > >> I'm testing attended transfer with 3 SIP phones. I noticed about 10% of >> my transfers make the call drop and I get this on my log: >> > > Some questions: > > 1. What asterisk version are you using? > 2. What are your SIP devices? > 3. Who is your SIP provider? (Judging by your CLI output, I'm guessing > you're using one.) > > > - Noah > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
