<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:courier,monaco,monospace,sans-serif;font-size:12pt">by the way thanks, I see you actually did file a bug so nevermind<br>that req.<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: Irshad Ahamed <irshad.fswitch@gmail.com><br>To: freeswitch-dev@lists.freeswitch.org; freeswitch-users@lists.freeswitch.org<br>Sent: Tuesday, January 8, 2008 11:53:35 AM<br>Subject: Re: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS....<br><br>
Your interpretation is right. This is what I have observed while testing here. To be more precise, when you hangup the FXS completely, the first call doesnt get disconnected.<br><br>BR,<br>Irshad.<br><br><div class="gmail_quote">
On Jan 8, 2008 7:38 PM, Anthony Minessale <<a rel="nofollow" ymailto="mailto:anthmct@yahoo.com" target="_blank" href="mailto:anthmct@yahoo.com">anthmct@yahoo.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;">
<div><div style="font-family: courier,monaco,monospace,sans-serif; font-size: 12pt;">Can you clarify more so I am sure I understand?<br><br>You use fxs to call an extension that leads to a sip phone.<br>You press FLASH on fxs to get a dialtone and make another call.
<br>You hangup the fxs completely (ONHOOK) and one of the 2 calls do not end?<br><br>The current intended behavior is when you FLASH fxs you can make<br>another call and switch back and forth with subsequent FLASH. But if
<br>the fxs is ONHOOK all the calls should end.<br><br>We still need some more code to do other behaviors from fxs like<br>transfer etc.<br><br><br><div> </div><div>Anthony Minessale II<br><br><span>FreeSWITCH <a rel="nofollow" target="_blank" href="http://www.freeswitch.org/">
http://www.freeswitch.org/</a></span><br><span>ClueCon <a rel="nofollow" target="_blank" href="http://www.cluecon.com/">http://www.cluecon.com/</a></span><br><br>AIM: anthm<br><a rel="nofollow" ymailto="mailto:MSN:anthony_minessale@hotmail.com" target="_blank" href="mailto:MSN:anthony_minessale@hotmail.com">
MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/PAYPAL:<a rel="nofollow" ymailto="mailto:anthony.minessale@gmail.com" target="_blank" href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a><br>IRC: <a rel="nofollow" target="_blank" href="http://irc.freenode.net">irc.freenode.net
</a> #freeswitch</div><div><br>FreeSWITCH Developer Conference<br><a rel="nofollow" ymailto="mailto:sip:888@conference.freeswitch.org" target="_blank" href="mailto:sip:888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a rel="nofollow" target="_blank" href="http://iax:guest@conference.freeswitch.org/888">
iax:guest@conference.freeswitch.org/888</a><br><a rel="nofollow" ymailto="mailto:googletalk:conf+888@conference.freeswitch.org" target="_blank" href="mailto:googletalk:conf+888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400</div><div><div></div><div class="Wj3C7c">
<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: "<a rel="nofollow" ymailto="mailto:freeswitch-dev@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-dev@lists.freeswitch.org">
freeswitch-dev@lists.freeswitch.org</a>" <<a rel="nofollow" ymailto="mailto:freeswitch-dev@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-dev@lists.freeswitch.org">freeswitch-dev@lists.freeswitch.org</a>><br>To: <a rel="nofollow" ymailto="mailto:freeswitch-dev@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-dev@lists.freeswitch.org">
freeswitch-dev@lists.freeswitch.org</a>; <a rel="nofollow" ymailto="mailto:freeswitch-users@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>Sent: Monday, January 7, 2008 11:36:09 PM<br>Subject: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS....
<br><br>
Set-up is as below:<br>1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXS + 1 FXO registered with SIPX.<br>2. SNOM300 registered with SIPX<br><span style="color: rgb(0, 0, 153);"><br> SNOM300:<br>
Kernel Version:snom300 linux 3.25</span><br style="color: rgb(0, 0, 153);">
<span style="color: rgb(0, 0, 153);"> Application-Version:snom300-SIP 6.5.2</span><br style="color: rgb(0, 0, 153);">
<span style="color: rgb(0, 0, 153);"> Rootfs-Version:snom300 jffs2 v3.36</span><br style="color: rgb(0, 0, 153);">
<span style="color: rgb(0, 0, 153);"> Firmware-URL:</span><br style="color: rgb(0, 0, 153);">
<span style="color: rgb(0, 0, 153);"> Production Information:Mac:00041325329C;Version:Standard;Hardware:snom300 (MB V3.2_A11);Date:26/06/06;Copyright(C) snom technology AG</span><br>3. GRANDSTREAM registered with SIPX.
<br><br><font color="#000099">GRANDSTREAM:
<br> Product Model: GXP2000<br> Software : Program-- <a rel="nofollow" target="_blank" href="http://1.1.4.18/">1.1.4.18</a><br> Version : </font><span style="color: rgb(0, 0, 153);">
Bootloader-- <a rel="nofollow" target="_blank" href="http://1.1.4.6/">
1.1.4.6</a><br></span><span style="color: rgb(0, 0, 153);"></span><br><br>Procedure followed:<br>1.
Make a successful call from Freeswitch FXS to SNOM300. Put the call on
hold @ Freeswitch FXS and make new call to GRANDSTREAM.
<br>2. Accept the call @ GRANDSTREAM and disconnect the connected call @ FREESWITCH FXS.<br>3. The second call with GRANDSTREAM is being disconnected but the first call with SNOM300 is still alive at SNOM300.<br><br>NOTE: Please find the Freeswitch logs and Ethereal caps @
<a rel="nofollow" target="_blank" href="http://jira.freeswitch.org/browse/MODENDP-74">http://jira.freeswitch.org/browse/MODENDP-74</a><br><br>BR,<br><font color="#888888">Irshad.</font>
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