<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:courier,monaco,monospace,sans-serif;font-size:12pt">What is it that you are calling in that example?<br>Is it a pstn gateway or something?<br><br>It sounds like the gateway you are calling always answers the call before it tries to dial the far end destination and does not have proper supervision handling.<br><br>The correct method would be that the gateway places the far end call without answering it and passes the indications across as they are encountered by the far end.<br><br>You may want to send your question to whoever is in charge of that gateway as I am fairly certain we do basic call scenarios correctly since we have had people interop with several major carriers.<br><br><br><div> </div><div>Anthony Minessale II<br><br><span>FreeSWITCH <a target="_blank"
href="http://www.freeswitch.org/">http://www.freeswitch.org/</a></span><br><span>ClueCon <a target="_blank" href="http://www.cluecon.com/">http://www.cluecon.com/</a></span><br><br>AIM: anthm<br>MSN:anthony_minessale@hotmail.com<br>GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<br>IRC: irc.freenode.net #freeswitch</div><div><br>FreeSWITCH Developer Conference<br>sip:888@conference.freeswitch.org<br>iax:guest@conference.freeswitch.org/888<br>googletalk:conf+888@conference.freeswitch.org<br>pstn:213-799-1400</div><div style="font-family: courier,monaco,monospace,sans-serif; font-size: 12pt;"><br><br><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;">----- Original Message ----<br>From: Steve Xu <516888@gmail.com><br>To: freeswitch-users@lists.freeswitch.org<br>Sent: Tuesday, March 4, 2008 6:01:56 PM<br>Subject: [Freeswitch-users] Does dialplan. bridge application can detect call hangup earlier?<br><br>
Hi all,<br><br>Here is another question. <br> I try to use dialplan<br><action application="bridge" data="<a rel="nofollow" ymailto="mailto:sofia/default/999917728693819@66.55.44.33" target="_blank" href="mailto:sofia/default/999917728693819@66.55.44.33">sofia/default/999917728693819@66.55.44.33</a>"/> <br>
to bridge two SIP endpoints. <br><br>Here are the strange pattern I found during call handling.<br><br>1.The called party still keep ringing for a long time period even I have hanged up the calling party already. I have to pick up the called party to stop the ring sometimes, however the call record shows the call has been answered. I think FS will detect the calling party status and drop the call if calling party hung up. It does not need to waiting for the status of called party changes.<br>
2.The calling party does not hear anything ( ie. a busy tone) when the called party hang up the call during the normal conversation, the call keep connecting there until the calling party hang up, then FS start realize the call ended.<br>
<br>Is there some configuration setting can make FS to detect the call status earlier and make the call handling properly?<br>or may be I used the wrong action application (bridge)to handle the calls?<br><br>Thanks again,<br>
<br>Steven Xu<br> <br><br>
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