Alejandro Recarey schrieb: > Doug, thanks for the help, already looked it up, but it does not seem > to be a NAT issue (which is what most posters suggest when googling) > > Danny, those are billsec durations, the call has been established and > media is being passed for 20 seconds. > > Thanks again! > > Alex > > Hi,
How do you dial the users? direct with the peername or something like ex...@ipofpeer ? i know this problem when dialing a patton ISDN ata without an extension. The call is established but when the T1 sip timeout fires the call gets disconnected. Maybe you could do some sip debugging and watch for resend sip messages. best regards steve -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // ------------------------------------------------- -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users