are you on older REV? try answering first to compare.
On Tue, Dec 1, 2009 at 6:29 PM, Esben Stien <[email protected]> wrote: > Anthony Minessale <[email protected]> writes: > > > do you want to generate ringback forever or no? > > Yes, I want the dialing party to get a ring forever. This is because I > cannot transfer the party to my SIP phone, because my SIP phone is > broken for incoming calls. > > I'll solve it by letting the party hear a ring and then I'll use the > event socket to give me a notification. I'll then dial in with my SIP > phone to a conference room and transfer the calling party there. > > I did this: > > <extension name="ring_forever"> > <condition field="destination_number" expression="ring_forever"> > <action application="gentones" data="${us-ring}|100"/> > <action application="transfer" data="ring_forever"/> > </condition> > </extension> > > <extension name="eternal_ring"> > <condition field="destination_number" expression="4ever"> > <action application="pre_answer"/> > <action application="transfer" data="ring_forever"/> > </condition> > </extension> > > ..which gives me an eternal ring, but it sounds choppy and it sounds as > if the following ring starts on top of the currently playing ring. > > I hear crackling sounds, which I don't experience with any other use of > freeswitch. > > Any idea? > > -- > Esben Stien is b...@e s a > http://www. s t n m > irc://irc. b - i . e/%23contact > sip:b0ef@ e e > jid:b0ef@ n n > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > 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:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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