The ringing is not passed across until the other phone (the one you are calling) sends a 180 Ringing.<br><br>As soon as it sends it, we pass that indication to the calling phone (your phone). <br><br>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:<br>
<br>go to line 145 of default.xml and put the following line as the first anti-action<br><anti-action application="ring_ready"/><br><br>This will make FreeSWITCH send your calling phone a 180 ringing before it knows for sure if it should.<br>
<br><br><br><div class="gmail_quote">On Tue, Apr 22, 2008 at 8:53 AM, Luis Jimenez <<a href="mailto:ljjimenez@gmail.com">ljjimenez@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Ok, my network topology is:<br><br>1 server HP ML-110 FS installed.<br>2 Snom 360<br>1 Switch Linksys SRW224P<br>using default dialplan installed by make samples<br><br>this is the debug of de FS console when you dial from 1000 to 1001:<br>
<br><br>freeswitch@pbx> 2008-04-22 08:33:43 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/<a href="mailto:1000@10.0.0.100" target="_blank">1000@10.0.0.100</a> [eddcc31a-3dcb-4644-8258-fcb5f49e6900]<br>
2008-04-22 08:33:43 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Juan Perez->1001@default<br>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<br>
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<br>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<br>
2008-04-22 08:33:45 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@10.0.0.16:2051;line=1nepvhw6 [dfaf0d12-a892-46f1-bfb1-1b0c385e97a5]<br>2008-04-22 08:33:45 [NOTICE] sofia.c:1713 sofia_handle_sip_i_state() Ring-Ready sofia/default/1001@10.0.0.16:2051;line=1nepvhw6!<br>
2008-04-22 08:33:45 [NOTICE] mod_sofia.c:1018 sofia_receive_message() Ring-Ready sofia/default/<a href="mailto:1000@10.0.0.100" target="_blank">1000@10.0.0.100</a>!<br>2008-04-22 08:33:45 [NOTICE] switch_ivr_originate.c:1036 switch_ivr_originate() Ring Ready sofia/default/<a href="mailto:1000@10.0.0.100" target="_blank">1000@10.0.0.100</a>!<br>
<br>Ring starts after last line<br><br>Any help appreciated.<br><font color="#888888">Luis jimenez</font><div><div></div><div class="Wj3C7c"><br><br><br><br><div class="gmail_quote">On Mon, Apr 21, 2008 at 6:28 AM, Brian West <<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I haven't seen this issue can you describe your network topology?<br>
<div><br>
On Apr 21, 2008, at 5:23 AM, Luis Jimenez wrote:<br>
<br>
> Ok, when you dial from say 1000 to 1001 you wait 3 seconds before<br>
> the phone start ringing nad you listen ringback, i'll send some<br>
> debugs from the FS console later.<br>
><br>
<br>
</div><div><div></div><div>Brian West<br>
<a href="mailto:sip%3Abrian@freeswitch.org" target="_blank">sip:brian@freeswitch.org</a><br>
<br>
<br>
<br>
<br>
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