We have a box running 1.6.2.11 on CentOS 5 using the RPM's from the Digium CentOS repository. We just left a 60 second voicemail on the system and had the full audio as well in the inbox. Not sure how your SIP configuration ties your SBC in, but native "users" created via users.conf and sip.conf appears to be working for me. Wouldnt be able to test more without knowing what settings you had between Asterisk and the SBC.
-- Trevor Benson dCAP, LPIC-1, CLA, Network+, MCP, CNA A1 Networks - Network Engineer DID (707)703-1041 FAX (707)703-1983 On Aug 26, 2010, at 8:47 AM, Steven C. Blair wrote: > > As a test we built Asterisk v1.6.2.11 on a new server. This version of > Asterisk exhibits the same behavior. From ngrep’s perspective we see an ACK > followed immediately by a BYE message. The user hears the recording being > played, begins to leave a message and is disconnected about 10 seconds into > the call. > > > > From: [email protected] > [mailto:[email protected]] On Behalf Of Steven C. Blair > Sent: Wednesday, August 25, 2010 2:08 PM > To: [email protected] > Subject: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question > > > We’re running Asterisk 1.6.1.17 for our campus voicemail server and Juniper > M120s as our SBC. Unanswered calls, which arrive via the SBC, are diverted to > voicemail using a 302 redirect when the called party doesn’t answer. In this > case the caller is able to hear the greetings and begin to leave a message > only to have Asterisk terminate the call mid-recording. > > We’re uncertain why this is happening and this is where we are hoping you > can help. In our environment the caller is any set on the PSTN. They call one > of our IP phones which no one answers. Our proxy, SER, responds to the SBC > with a 302 redirect and the call is diverted to Asterisk. The caller hears > the unavailable greeting for 6-4050, begins to leave a message and is cut-off > after about 10 seconds. In an ngrep trace we see Asterisk receive an ACK from > the SBC and it immediately responds with a BYE message for that call. > > Has anyone else experienced this type of issue? > > > --- > > ISC Networking & Telecommunications > 3401 Walnut Street, Suite 221A > Philadelphia, PA 19104 > 215-573-8396 > 215-898-9348 (fax) > > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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
