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

[Ryan Wagoner]

On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg <luke at 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-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


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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