On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg <l...@solvent-llc.com> wrote:
> Doug: > for what it's worth I am having the exact same nightmare. Not sure exactly > when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I > am > running 1.8.9rc1). I also have Polycom (335, 550, 650) and blind > transfers > are broken. All legs of the call are dropped when the xfer is executed. A > calls B, B xfer to C and (C) blips for a split second like its ringing but > then all calls go dead. I tried to debug myself using some sip tracing but > I didn't get very far. I even tried mucking around with a few settings in > my Polycom provisioning I thought might be related e.g. > > voIpProt.SIP.allowTransferOnProceeding > voIpProt.SIP.connectionReuse.useAlias > voIpProt.SIP.useContactInReferTo > voIpProt.SIP.conference.parallelRefer > voIpProt.SIP.strictLineSeize > voIpProt.SIP.strictUserValidation > voIpProt.SIP.strictReplacesHeader > voIpProt.SIP.useContactInReferTo > > and also upgraded to the new 3.3.4 firmware which is out yesterday, didn't > change a thing. > stuck here for now, Attended xfers seem to work. I am not sure this is > a > Polycom-specific issue because I was seeing this bad behavior even using > some Softphones I set up for testing. > > my next recourse is to try rolling back to 1.8.8.0 or earlier and if that > fixes it then I will open a JIRA ticket with more details. > > Luke > > > -- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas > Mortensen > Sent: Thursday, January 05, 2012 3:04 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Blind transfers being cancelled by asterisk & > hanging up on remote caller > > Hello all, > > I have a system running AsteriskNOW with asterisk > asterisk-1.8.8.1-1_centos5 > from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so > that > blindpreferred=1 (all transfers default as blind transfers). If a customer > calls in & we answer & transfer, everything works fine. But if we call out > to a customer & then transfer to another internal extension, that extension > quickly rings & then the call is immediately gone & hung up. We are using > Polycom firmware 3.3.3. > > In troubleshooting this & analyzing the asterisk logs (& asterisk SIP > debug), I am seeing a few interesting items. Any help would be appreciated. > > [...] > > Thanks, > - > Doug Mortensen 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. Ryan
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