[Freeswitch-users] Polycom Consultative Transfer and Voicemail

[Anthony Minessale]

if you used the loopback endpoint to loop around to voicemail or made a
looped sip call back to your own box you could xfer it as desired.


bridge to "loopback/app=voicemail:default ${domain_name}
${dialed_extension}"

That will make the vm app run as a channel instead of an inline app.

This is an undocumented feature because it's not well tested so if it
doesn't work *shrug* =D



On Thu, Jan 21, 2010 at 6:00 PM, Adam Ford <lists at redbonez.net> wrote:

> That link didn't come through very well, here is a shortened one -
> http://bit.ly/6wDAXD
>
> -Adam
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
> Ford
> Sent: Thursday, January 21, 2010 4:43 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
>
> Yes it is a known issue with Polycom phones. Polycom supports a
> non-standard
> transfer method which does not work with FreeSWITCH.
>
> See this article for further details -
>
> http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic
> es/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth<http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic%0Aes/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth>
>
> I ran into the same problem, disabling
> voIpProt.SIP.allowTransferOnProceeding as suggested in that article
> resolved
> the issue for me.
>
> -Adam
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Troy
> Anderson
> Sent: Thursday, January 21, 2010 4:21 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail
>
> Hello,
>
> I'm on the latest trunk version (16440) and having an issue with Polycom
> and
> transferring.  The dial plan is set up so that unanswered calls go to
> voicemail.  When I answer a call with a polycom phone and then transfer
> that
> call to another phone, if the other phone doesn't pick up and the voicemail
> app starts, then I hit transfer again with the intent of having the caller
> leave a voicemail, the call is dropped.  If the phone does pick up during
> the transfer, it works fine.
>
> I also have an Aastra phone, and when I do the same thing, but from the
> Aastra phone, it works as expected.  Is this known to be a problem with
> Polycom?
>
> Thanks!
> Troy
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100121/6a0c1870/attachment-0002.html