GSM device or not, just determine what line disconnect signalling is used in each port. the GWX gateway must be set to detect this. otherwise, there's no way FS can hangup the call. this signal must be present when using the SIP gateway.<br>
<br>also test these 2 scenarios - 1) FXO1 hangs up first; 2) FXO2 hangs up first.<br><br><div class="gmail_quote">On Sun, Apr 15, 2012 at 12:48 AM, ocset <span dir="ltr"><<a href="mailto:ocset@the800group.com">ocset@the800group.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
Thanks Nandy<br>
<br>
I am not using the GSM device in this test and have tried changing
the disconnect signalling without any luck. When I hang up the
phones, nothing is recorder in the Freeswitch log (log level 7). Is
there a way to trap the disconnect signal by writing a "lua" script
or would that info already be written to the log if it was being
received?.<br>
<br>
What I don't understand is that when I use the SIP gateway for the
outgoing leg of the call, then the disconnect signal from the two
phones is received. But when I use the PSTN line as the outgoing
leg, the disconnect signal is not received from either phone?<br>
<br>
Anyone got any suggestions how I would debug this?<br>
<br>
Thanks again for your help.<span class="HOEnZb"><font color="#888888"><br>
O</font></span><div><div class="h5"><br>
<br>
<br>
On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
<blockquote type="cite">check the line supervision signalling of your GSM and
the GXW4104. there are several disconnect signalling used: current
disconnect, polarity reversal or busy tones. your FXO port must be
set accordingly. <br>
<br>
<div class="gmail_quote">
On Fri, Apr 13, 2012 at 3:09 PM, ocset <span dir="ltr"><<a href="mailto:ocset@the800group.com" target="_blank">ocset@the800group.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi<br>
<br>
I have successfully managed to get FreeSwitch to bridge an
incoming call<br>
on a PSTN line to an external mobile phone using a second PSTN
line with<br>
the help of a GXW4104 FSO gateway.<br>
<br>
I am now having a serious issue where the call never gets
terminated<br>
when the parties hang up. The only way to end the call is to
either<br>
shut-down FreeSwitch or turn off the GXW4104. If I bridge the
call to a<br>
SIP gateway (PennyTel) instead of the second PSTN line, then
it works as<br>
expected when both parties hang up.<br>
<br>
Here is what I have tested<br>
<br>
1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway
-> external<br>
mobile phone -- call end when both parties hang up.<br>
2. Call -> PSTN line1 (GXW4104) -> PSTN line2 (GXW4104)
-> external<br>
mobile phone -- call never end, even when both parties hang
up.<br>
<br>
Is there a parameter I should be setting to help solve this
issue or<br>
some code that I should implement to test for this situation?<br>
<br>
I have also noticed that with test scenario 1 (PSTN line + SIP
Gateway),<br>
if the person being called does not end the call on their
side, then<br>
that leg of the bridge remains active.<br>
<br>
ps. The second PSTN line is not a real copper line. It is a
PennyTel<br>
VOIP line that is managed by the modem and connected to the
GXW4104 via<br>
phone cable. So, as far as the GXW4104 is concerned, it is
just another<br>
POTS line. I don't have a second PSTN phone line so can't test
any other<br>
way.<br>
<br>
Thanks in advance<br>
O.<br>
<br>
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