[asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller

[Luke Hamburg]

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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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