[Freeswitch-users] [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS....

[Anthony Minessale]

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 at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400


----- Original Message ----
From: Irshad Ahamed <irshad.fswitch at gmail.com>
To: freeswitch-dev at lists.freeswitch.org; freeswitch-users at 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 at 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 at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net
 #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org

iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400




----- Original Message ----
From: "
freeswitch-dev at lists.freeswitch.org" <freeswitch-dev at lists.freeswitch.org>
To: 
freeswitch-dev at lists.freeswitch.org; freeswitch-users at 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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