I get the idea, but can't seem to get it to work. I tried doing a bridge to "loopback/app=bridge ${sofia_contact(${dialed_user}@${dialed_domain}", but fs complained => Cannot create outgoing channel of type [loopback=app:sofia] Also, I tried "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. I tried the suggestions about modifying the sip.cfg for the phones and that does work (thanks!), but that forces a Consultative Transfer. 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. Thanks, Troy On Jan 21, 2010, at 5:48 PM, Anthony Minessale wrote:
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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Adam Ford Sent: Thursday, January 21, 2010 4:43 PM To: freeswitch-users@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
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Troy Anderson Sent: Thursday, January 21, 2010 4:21 PM To: freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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