<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div>I get the idea, but can't seem to get it to work.</div><div><br></div><div>I tried doing a&nbsp;bridge to "loopback/app=bridge&nbsp;${sofia_contact(${dialed_user}@${dialed_domain}", but fs complained =&gt; Cannot create outgoing channel of type [loopback=app:sofia]</div><div>Also, I tried&nbsp;"loopback/app=voicemail:default ${domain_name} ${dialed_extension}" and it did get to voicemail, but it didn't prompt me for any info - just immediately complained that the recording was too short.</div><div><br></div><div>I tried the suggestions about modifying the sip.cfg for the phones and that does work (thanks!), but that forces a Consultative Transfer. &nbsp;It would be nice to get this method working, which would result in the transfer button doing a Consultive Transfer unless you hang up, then it would be like a blind transfer.</div><div><br></div><div>Thanks,</div><div>Troy</div><div><br><div><div>On Jan 21, 2010, at 5:48 PM, Anthony Minessale wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">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.<br><br><br>bridge to "loopback/app=voicemail:default ${domain_name} ${dialed_extension}"<br>
<br>That will make the vm app run as a channel instead of an inline app.<br><br>This is an undocumented feature because it's not well tested so if it doesn't work *shrug* =D<br><br><br><br><div class="gmail_quote">
On Thu, Jan 21, 2010 at 6:00 PM, Adam Ford <span dir="ltr">&lt;<a href="mailto:lists@redbonez.net">lists@redbonez.net</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
That link didn't come through very well, here is a shortened one -<br>
<a href="http://bit.ly/6wDAXD" target="_blank">http://bit.ly/6wDAXD</a><br>
<div class="im"><br>
-Adam<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><br>
</div><div class="im">[mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of Adam<br>
Ford<br>
Sent: Thursday, January 21, 2010 4:43 PM<br>
To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
</div><div><div></div><div class="h5">Subject: Re: [Freeswitch-users] Polycom Consultative Transfer and Voicemail<br>
<br>
Yes it is a known issue with Polycom phones. Polycom supports a non-standard<br>
transfer method which does not work with FreeSWITCH.<br>
<br>
See this article for further details -<br>
<a href="http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic%0Aes/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth" target="_blank">http://www.junctionnetworks.com/knowledgebase/onsip/phones-routers-and-devic<br>

es/phone-configuration/polycom/polycom-disabling-non-standard-transfer-meth</a><br>
<br>
I ran into the same problem, disabling<br>
voIpProt.SIP.allowTransferOnProceeding as suggested in that article resolved<br>
the issue for me.<br>
<br>
-Adam<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><br>
[mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of Troy<br>
Anderson<br>
Sent: Thursday, January 21, 2010 4:21 PM<br>
To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Subject: [Freeswitch-users] Polycom Consultative Transfer and Voicemail<br>
<br>
Hello,<br>
<br>
I'm on the latest trunk version (16440) and having an issue with Polycom and<br>
transferring. &nbsp;The dial plan is set up so that unanswered calls go to<br>
voicemail. &nbsp;When I answer a call with a polycom phone and then transfer that<br>
call to another phone, if the other phone doesn't pick up and the voicemail<br>
app starts, then I hit transfer again with the intent of having the caller<br>
leave a voicemail, the call is dropped. &nbsp;If the phone does pick up during<br>
the transfer, it works fine.<br>
<br>
I also have an Aastra phone, and when I do the same thing, but from the<br>
Aastra phone, it works as expected. &nbsp;Is this known to be a problem with<br>
Polycom?<br>
<br>
Thanks!<br>
Troy<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
Twitter: <a href="http://twitter.com/FreeSWITCH_wire">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net/">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br>
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