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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Are we the only 2 people on this list experiencing this issue? (surprised)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Anyone else have any insights?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>My hunch is that this is likely some type of FreePBX issue with how it generates the [from-internal-xfer] context.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Douglas Mortensen<br><b>Sent:</b> Saturday, January 07, 2012 3:16 PM<br><b>To:</b> asterisk-users@lists.digium.com<br><b>Subject:</b> Re: [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'>Oh crap. I just reread the previous post & realized I'm not alone. Hallelujah! I'll post back more info soon.<br><br>-<br>Doug Mortensen <br></span><i><span style='font-size:10.5pt;font-family:"Arial","sans-serif";color:#333333'>Sent via DroidX2 on Verizon Wireless™</span></i><span style='font-family:"Arial","sans-serif"'><o:p></o:p></span></p></div><p class=MsoNormal><br><br>-----Original message-----<o:p></o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'><b><span style='font-size:10.5pt;font-family:"Arial","sans-serif"'>From: </span></b><span style='font-size:10.5pt;font-family:"Arial","sans-serif"'>Ryan Wagoner <<a href="mailto:rswagoner@gmail.com">rswagoner@gmail.com</a>><b><br>To: </b>Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><b><br>Sent: </b>Sat, Jan 7, 2012 15:59:36 GMT+00:00<b><br>Subject: </b>Re: [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller<o:p></o:p></span></p></div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg <<a href="mailto:luke@solvent-llc.com">luke@solvent-llc.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Doug:<br>for what it's worth I am having the exact same nightmare. Not sure exactly<br>when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I am<br>running 1.8.9rc1). I also have Polycom (335, 550, 650) and blind transfers<br>are broken. All legs of the call are dropped when the xfer is executed. A<br>calls B, B xfer to C and (C) blips for a split second like its ringing but<br>then all calls go dead. I tried to debug myself using some sip tracing but<br>I didn't get very far. I even tried mucking around with a few settings in<br>my Polycom provisioning I thought might be related e.g.<br><br> voIpProt.SIP.allowTransferOnProceeding<br> voIpProt.SIP.connectionReuse.useAlias<br> voIpProt.SIP.useContactInReferTo<br> voIpProt.SIP.conference.parallelRefer<br> voIpProt.SIP.strictLineSeize<br> voIpProt.SIP.strictUserValidation<br> voIpProt.SIP.strictReplacesHeader<br> voIpProt.SIP.useContactInReferTo<br><br>and also upgraded to the new 3.3.4 firmware which is out yesterday, didn't<br>change a thing.<br>stuck here for now, Attended xfers seem to work. I am not sure this is a<br>Polycom-specific issue because I was seeing this bad behavior even using<br>some Softphones I set up for testing.<br><br>my next recourse is to try rolling back to 1.8.8.0 or earlier and if that<br>fixes it then I will open a JIRA ticket with more details.<br><br>Luke<br><br><br>--<br>From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Douglas<br>Mortensen<br>Sent: Thursday, January 05, 2012 3:04 PM<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>Subject: [asterisk-users] Blind transfers being cancelled by asterisk &<br>hanging up on remote caller<o:p></o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'><br>Hello all,<br><br>I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5<br>from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that<br>blindpreferred=1 (all transfers default as blind transfers). If a customer<br>calls in & we answer & transfer, everything works fine. But if we call out<br>to a customer & then transfer to another internal extension, that extension<br>quickly rings & then the call is immediately gone & hung up. We are using<br>Polycom firmware 3.3.3.<br><br>In troubleshooting this & analyzing the asterisk logs (& asterisk SIP<br>debug), I am seeing a few interesting items. Any help would be appreciated.<o:p></o:p></p></div><p class=MsoNormal>[...]<br><br>Thanks,<br>-<br>Doug Mortensen<o:p></o:p></p><div><p class=MsoNormal><br>I can't reproduce this on a test system with Asterisk 1.8.8.1 using a Polycom 335 and 550 running firmware 3.2.6. I called an external number using Vitelity then blind transferred to the other phone. I am interested as I have a production system with Polycom 335 phones running 1.8.7.0 that works.<br><br>Ryan<o:p></o:p></p></div></div></div></div></body></html>