<html><head><base href="x-msg://20/"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">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.<div><br></div><div><br><div>
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div>--</div><div>Trevor Benson</div><div>dCAP, LPIC-1, CLA, Network+, MCP, CNA</div><div><div>A1 Networks - Network Engineer</div></div><div>DID (707)703-1041</div><div>FAX (707)703-1983</div><div><br></div></div></span><br class="Apple-interchange-newline"></div></span><br class="Apple-interchange-newline"></div></span><br class="Apple-interchange-newline"></span><br class="Apple-interchange-newline">
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<br><div><div>On Aug 26, 2010, at 8:47 AM, Steven C. Blair wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; font-family: Helvetica; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div lang="EN-US" link="blue" vlink="purple"><div class="WordSection1" style="page: WordSection1; "><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span style="color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span style="color: rgb(31, 73, 125); ">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.<o:p></o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span style="color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span style="color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><span style="color: rgb(31, 73, 125); "><o:p> </o:p></span></div><div><div style="border-right-style: none; border-bottom-style: none; border-left-style: none; border-width: initial; border-color: initial; border-top-style: solid; border-top-color: rgb(181, 196, 223); border-top-width: 1pt; padding-top: 3pt; padding-right: 0in; padding-bottom: 0in; padding-left: 0in; "><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><b><span style="font-size: 10pt; font-family: Tahoma, sans-serif; ">From:</span></b><span style="font-size: 10pt; font-family: Tahoma, sans-serif; "><span class="Apple-converted-space"> </span><a href="mailto:asterisk-users-bounces@lists.digium.com" style="color: blue; text-decoration: underline; ">asterisk-users-bounces@lists.digium.com</a><span class="Apple-converted-space"> </span>[mailto:asterisk-users-bounces@lists.digium.com]<span class="Apple-converted-space"> </span><b>On Behalf Of<span class="Apple-converted-space"> </span></b>Steven C. Blair<br><b>Sent:</b><span class="Apple-converted-space"> </span>Wednesday, August 25, 2010 2:08 PM<br><b>To:</b><span class="Apple-converted-space"> </span><a href="mailto:asterisk-users@lists.digium.com" style="color: blue; text-decoration: underline; ">asterisk-users@lists.digium.com</a><br><b>Subject:</b><span class="Apple-converted-space"> </span>[asterisk-users] Asterisk 1.6.1.17 ACK/BYE question<o:p></o:p></span></div></div></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "> 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.<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "> 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.<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Has anyone else experienced this type of issue?<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">---<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">ISC Networking & Telecommunications<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">3401 Walnut Street, Suite 221A<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">Philadelphia, PA 19104<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">215-573-8396<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; ">215-898-9348 (fax)<o:p></o:p></div><div style="margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 11pt; font-family: Calibri, sans-serif; "><o:p> </o:p></div></div>--<span class="Apple-converted-space"> </span><br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by<span class="Apple-converted-space"> </span><a href="http://www.api-digital.com" style="color: blue; text-decoration: underline; ">http://www.api-digital.com</a><span class="Apple-converted-space"> </span>--<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" style="color: blue; text-decoration: underline; ">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" style="color: blue; text-decoration: underline; ">http://lists.digium.com/mailman/listinfo/asterisk-users</a></div></span></blockquote></div><br></div></body></html>