I tested with the latest svn revision but the same issue exists still. The SVN revision which I tested upon is ... Path: . URL: http://svn.freeswitch.org/svn/freeswitch/trunk Repository Root: http://svn.freeswitch.org/svn Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2 Revision: 7148 Node Kind: directory Schedule: normal Last Changed Author: anthm Last Changed Rev: 7148 Last Changed Date: 2008-01-09 04:00:52 +0530 (Wed, 09 Jan 2008) Properties Last Updated: 2008-01-08 11:14:19 +0530 (Tue, 08 Jan 2008) BR, Irshad. On Jan 9, 2008 5:05 AM, Anthony Minessale <anthmct@yahoo.com> wrote:
Please update the code to the latest revision where it should be working right.
In the future could you please direct issues to http://jira.freeswitch.org so we can properly track them.
thanks =)
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
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----- 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-users] [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.
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