What about it? Did you try my suggestion? I am not sure I could have explained it any better?
Watch your console trace and maybe take a pcap of it. you can export TPORT_LOG=1 in your shell to see the sip messages in the console. The instant the phone you are calling sends us 180, we send you 180. So if you add the line i told you to your dialplan, we will send 180 instantly instead which is what you want right? On Tue, Apr 22, 2008 at 9:18 AM, Luis Jimenez <[EMAIL PROTECTED]> wrote: > What about OpenSer, i get the same results as Asterisk. > > > > On Tue, Apr 22, 2008 at 10:11 AM, Anthony Minessale < > [EMAIL PROTECTED]> wrote: > > > The ringing is not passed across until the other phone (the one you are > > calling) sends a 180 Ringing. > > > > As soon as it sends it, we pass that indication to the calling phone > > (your phone). > > > > Asterisk just assumes you should hear ringing and sends it instantly on > > it's own before anyone knows that the call is going to work. If you want > > this same behaviour, add this to the dialplan: > > > > go to line 145 of default.xml and put the following line as the first > > anti-action > > <anti-action application="ring_ready"/> > > > > This will make FreeSWITCH send your calling phone a 180 ringing before > > it knows for sure if it should. > > > > > > > > > > On Tue, Apr 22, 2008 at 8:53 AM, Luis Jimenez <[EMAIL PROTECTED]> > > wrote: > > > > > Ok, my network topology is: > > > > > > 1 server HP ML-110 FS installed. > > > 2 Snom 360 > > > 1 Switch Linksys SRW224P > > > using default dialplan installed by make samples > > > > > > this is the debug of de FS console when you dial from 1000 to 1001: > > > > > > > > > [EMAIL PROTECTED]> 2008-04-22 08:33:43 [NOTICE] switch_channel.c:531 > > > switch_channel_set_name() New Channel sofia/default/[EMAIL PROTECTED] > > > 2008-04-22 08:33:43 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() > > > Processing Juan Perez->[EMAIL PROTECTED] > > > 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 > > > switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::dx XML > > > features > > > 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 > > > switch_ivr_bind_dtmf_meta_session() Bound: 2 > > > record_session::/opt/freeswitch/recordings/1000.2008-04-22-08-33-44.wav > > > 2008-04-22 08:33:44 [INFO] switch_ivr_async.c:1395 > > > switch_ivr_bind_dtmf_meta_session() Bound: 3 execute_extension::cf XML > > > features > > > 2008-04-22 08:33:45 [NOTICE] switch_channel.c:531 > > > switch_channel_set_name() New Channel sofia/default/[EMAIL > > > PROTECTED]:2051;line=1nepvhw6 > > > [dfaf0d12-a892-46f1-bfb1-1b0c385e97a5] > > > 2008-04-22 08:33:45 [NOTICE] sofia.c:1713 sofia_handle_sip_i_state() > > > Ring-Ready sofia/default/[EMAIL PROTECTED]:2051;line=1nepvhw6! > > > 2008-04-22 08:33:45 [NOTICE] mod_sofia.c:1018 sofia_receive_message() > > > Ring-Ready sofia/default/[EMAIL PROTECTED] > > > 2008-04-22 08:33:45 [NOTICE] switch_ivr_originate.c:1036 > > > switch_ivr_originate() Ring Ready sofia/default/[EMAIL PROTECTED] > > > > > > Ring starts after last line > > > > > > Any help appreciated. > > > Luis jimenez > > > > > > > > > > > > > > > On Mon, Apr 21, 2008 at 6:28 AM, Brian West <[EMAIL PROTECTED]> > > > wrote: > > > > > > > I haven't seen this issue can you describe your network topology? > > > > > > > > On Apr 21, 2008, at 5:23 AM, Luis Jimenez wrote: > > > > > > > > > Ok, when you dial from say 1000 to 1001 you wait 3 seconds before > > > > > the phone start ringing nad you listen ringback, i'll send some > > > > > debugs from the FS console later. > > > > > > > > > > > > > Brian West > > > > sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > 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 > > > > > > > > > > > > > _______________________________________________ > > > 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/ > > > > AIM: anthm > > MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > > iax:[EMAIL PROTECTED]/888 > > googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> > > pstn:213-799-1400 > > _______________________________________________ > > 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 > > > > > > _______________________________________________ > 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/ AIM: anthm MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> pstn:213-799-1400
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