by the way thanks, I see you actually did file a bug so nevermind that req. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Irshad Ahamed <irshad.fswitch@gmail.com> To: freeswitch-dev@lists.freeswitch.org; freeswitch-users@lists.freeswitch.org Sent: Tuesday, January 8, 2008 11:53:35 AM Subject: Re: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS.... 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, Irshad. On Jan 8, 2008 7:38 PM, Anthony Minessale <anthmct@yahoo.com> wrote: Can you clarify more so I am sure I understand? You use fxs to call an extension that leads to a sip phone. You press FLASH on fxs to get a dialtone and make another call. You hangup the fxs completely (ONHOOK) and one of the 2 calls do not end? The current intended behavior is when you FLASH fxs you can make another call and switch back and forth with subsequent FLASH. But if the fxs is ONHOOK all the calls should end. We still need some more code to do other behaviors from fxs like transfer etc. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: " freeswitch-dev@lists.freeswitch.org" <freeswitch-dev@lists.freeswitch.org> To: freeswitch-dev@lists.freeswitch.org; freeswitch-users@lists.freeswitch.org Sent: Monday, January 7, 2008 11:36:09 PM Subject: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS.... Set-up is as below: 1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXS + 1 FXO registered with SIPX. 2. SNOM300 registered with SIPX SNOM300: Kernel Version:snom300 linux 3.25 Application-Version:snom300-SIP 6.5.2 Rootfs-Version:snom300 jffs2 v3.36 Firmware-URL: Production Information:Mac:00041325329C;Version:Standard;Hardware:snom300 (MB V3.2_A11);Date:26/06/06;Copyright(C) snom technology AG 3. GRANDSTREAM registered with SIPX. GRANDSTREAM: Product Model: GXP2000 Software : Program-- 1.1.4.18 Version : Bootloader-- 1.1.4.6 Procedure followed: 1. Make a successful call from Freeswitch FXS to SNOM300. Put the call on hold @ Freeswitch FXS and make new call to GRANDSTREAM. 2. Accept the call @ GRANDSTREAM and disconnect the connected call @ FREESWITCH FXS. 3. The second call with GRANDSTREAM is being disconnected but the first call with SNOM300 is still alive at SNOM300. NOTE: Please find the Freeswitch logs and Ethereal caps @ http://jira.freeswitch.org/browse/MODENDP-74 BR, Irshad. Looking for last minute shopping deals? Find them fast with Yahoo! Search. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping