<p>It is what your telephone company's equipment does, not your mobile phone provider. </p>
<div class="gmail_quote">On Apr 15, 2012 12:39 PM, "ocset" <<a href="mailto:ocset@the800group.com">ocset@the800group.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
Hi Nandy<br>
<br>
Thanks for you help. <br>
<br>
Just to answer you question first. It makes not difference who hangs
up first, the calls never terminate.<br>
<br>
I have enabled "Polarity Reversal" on the GXW4104 and that has
solved the problem. What I don't get is why this would work. I
thought Polarity Reversal is old school and how would my mobile
phone be doing this or is this being done by the mobile phone
provider/carrier?<br>
<br>
I now have "Enable Current Disconnect", "Enable Tone Disconnect" and
"Enable Polarity Reversal" set for good measure even though the
"Reverse Polarity" was the one that seemed to fix the problem.<br>
<br>
Regards<br>
O.<br>
<br>
On 04/15/2012 09:01 AM, Nandy Dagondon wrote:
<blockquote type="cite">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" 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">
<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><font color="#888888"><br>
O</font></span>
<div>
<div><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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