perhaps, your dialtone parameters are wrong causing the problem. try to enable it w/ the right parameters.<br>-nandy<br>
<br><br><div class="gmail_quote">On Tue, Sep 20, 2011 at 8:06 PM, ocset <span dir="ltr"><<a href="mailto:ocset@the800group.com">ocset@the800group.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div bgcolor="#FFFFFF" text="#000000">
I fixed the problem.<br>
<br>
The guide I followed for setting up the incoming calls suggested
this setting: <font color="#000099">"Wait for Dial-Tone(Y/N):
ch1-4:Y;"</font><br>
<br>
I changed this to "N" and now it works. Not sure how that affects
the use of the system but at least incoming call are now working.<br>
<br>
Thanks for all your help.<div><div></div><div class="h5"><br>
<br>
<br>
On 09/16/2011 03:35 PM, Nandy Dagondon wrote:
<blockquote type="cite">pls see my comments below. you're welcome.<br>
-nandy<br>
<br>
<div class="gmail_quote">On Fri, Sep 16, 2011 at 1:14 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: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div bgcolor="#FFFFFF" text="#000000"> Hi Nandy<br>
<br>
Thanks for your help so far - unfortunately this is still
not working.<br>
<br>
Based on your examples, I have created the following two
files <br>
<br>
1. <br>
../sip_profile/internal/01_custom.xml<br>
<br>
<include><br>
<param name="multiple-registrations" value="true"/><br>
<param name="accept-blind-reg" value="true"/><br>
<param name="accept-blind-auth" value="true"/><br>
<br>
<gateway name="gxw4104-fxo">
<div><br>
<param name="username" value="1019"/><br>
</div>
<div> <param name="password"
value="1234"/><br>
</div>
<param name="realm" value="192.168.0.160"/><br>
<param name="sip-port" value="5060"/><br>
<param name="rtp_ip" value="192.168.0.160"/>
<div><br>
<param name="dtmf-type" value="rfc2833"/><br>
<param name="expire-seconds" value="600"/><br>
<param name="register" value="false"/><br>
<param name="caller-id-in-from" value="false"/><br>
</gateway><br>
</include><br>
<br>
</div>
2.<br>
../dialplan/default/01_custom.xml<br>
<br>
<include><br>
<extension name="gxw4104-fxo-local">
<div><br>
<condition field="${toll_allow}"
expression="local"/><br>
</div>
<condition field="destination_number"
expression="^(\d{10})$"><br>
<action application="set"
data="effective_caller_id_number=5555555555"/>
<div><br>
<action application="set"
data="effective_caller_id_name=ThisIsMyCompany"/><br>
<action application="set"
data="ignore_early_media=ring_ready"/><br>
<action application="set"
data="ringback=${us-ring}"/><br>
</div>
<action application="bridge" data=<a href="mailto:sofia/gateway/gxw4104-fxo/$1@192.168.0.160:5060" target="_blank">"sofia/gateway/gxw4104-fxo/$1@192.168.0.160:5060"</a>/><br>
</condition><br>
</extension><br>
</include> <br>
<br>
<br>
<br>
Command "sofia status" shows the followiing<br>
<br>
internal profile
<a href="mailto:sip:mod_sofia@192.168.0.23:5060" target="_blank">sip:mod_sofia@192.168.0.23:5060</a>
RUNNING (0)<br>
internal::gxw4104-fxo gateway <a href="mailto:sip:1019@192.168.0.160" target="_blank">sip:1019@192.168.0.160</a>
NOREG<br>
external profile
<a href="mailto:sip:mod_sofia@192.168.0.23:5080" target="_blank">sip:mod_sofia@192.168.0.23:5080</a>
RUNNING (0)<br>
external::<a href="http://example.com" target="_blank">example.com</a>
gateway <a href="mailto:sip:joeuser@example.com" target="_blank">sip:joeuser@example.com</a>
NOREG<br>
internal-ipv6 profile
sip:mod_sofia@[::1]:5060 RUNNING (0)<br>
192.168.0.23 alias
internal
ALIASED<br>
<br>
<br>
Here is the call log when I try do dial out:<br>
<br>
<a href="tel:2011-09-16" value="+6320110916" target="_blank">2011-09-16</a>
12:57:30.903616 [NOTICE] switch_channel.c:669 New Channel <a href="mailto:sofia/internal/1014@192.168.0.23" target="_blank">sofia/internal/1014@192.168.0.23</a>
[0a83e803-b3fc-4a46-bc74-9d6786dac8e2]<br>
<a href="tel:2011-09-16" value="+6320110916" target="_blank">2011-09-16</a>
12:57:30.933863 [INFO] mod_dialplan_xml.c:418 Processing
User 1014->0412345678 in context default<br>
<a href="tel:2011-09-16" value="+6320110916" target="_blank">2011-09-16</a>
12:57:30.950359 [NOTICE] switch_channel.c:669 New Channel <a href="mailto:sofia/internal/0412345678@192.168.0.160:5060" target="_blank">sofia/internal/0412345678@192.168.0.160:5060</a>
[24111b8d-25f0-4433-a49f-88973730ebfb]<br>
<a href="tel:2011-09-16" value="+6320110916" target="_blank">2011-09-16</a>
12:57:40.957391 [NOTICE] sofia.c:4789 Hangup <a href="mailto:sofia/internal/0412345678@192.168.0.160:5060" target="_blank">sofia/internal/0412345678@192.168.0.160:5060</a>
[CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]<br>
<a href="tel:2011-09-16" value="+6320110916" target="_blank">2011-09-16</a>
12:57:40.958554 [NOTICE] switch_core_session.c:1182 Session
7 (<a href="mailto:sofia/internal/0412345678@192.168.0.160:5060" target="_blank">sofia/internal/0412345678@192.168.0.160:5060</a>)
Ended<br>
<a href="tel:2011-09-16" value="+6320110916" target="_blank">2011-09-16</a>
12:57:40.958554 [NOTICE] switch_core_session.c:1184 Close
Channel <a href="mailto:sofia/internal/0412345678@192.168.0.160:5060" target="_blank">sofia/internal/0412345678@192.168.0.160:5060</a>
[CS_DESTROY]<br>
<a href="tel:2011-09-16" value="+6320110916" target="_blank">2011-09-16</a>
12:57:40.958554 [INFO] mod_dptools.c:2355 Originate Failed.
Cause: NORMAL_TEMPORARY_FAILURE<br>
<a href="tel:2011-09-16" value="+6320110916" target="_blank">2011-09-16</a>
12:57:40.958554 [NOTICE] mod_dptools.c:2418 Hangup <a href="mailto:sofia/internal/1014@192.168.0.23" target="_blank">sofia/internal/1014@192.168.0.23</a>
[CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]<br>
<a href="tel:2011-09-16" value="+6320110916" target="_blank">2011-09-16</a>
12:57:40.995191 [NOTICE] switch_core_session.c:1182 Session
6 (<a href="mailto:sofia/internal/1014@192.168.0.23" target="_blank">sofia/internal/1014@192.168.0.23</a>)
Ended<br>
<a href="tel:2011-09-16" value="+6320110916" target="_blank">2011-09-16</a>
12:57:40.995191 [NOTICE] switch_core_session.c:1184 Close
Channel <a href="mailto:sofia/internal/1014@192.168.0.23" target="_blank">sofia/internal/1014@192.168.0.23</a>
[CS_DESTROY]<br>
<br>
<br>
<br>
I have the following questions<br>
<br>
1. What is the significance of the user 1019? I have a
default install of FS so that user does exist but I am not
logged in as that user on my sip phone. I am logged in as
user 1014.<br>
<br>
</div>
</blockquote>
<div>1019 is the user account for the FXO port to handle
incoming calls. so, the login details must be set on
directory/default/1019.xml. <br>
<br>
actually, the user and password entries can be deleted
because FS is not registering to the FXO gateway.<br>
<br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div bgcolor="#FFFFFF" text="#000000"> 2. The resultant log
does not show my gateway being used but instead shows <a href="mailto:/sofia/internal/0412345678@192.168.0.160:5060" target="_blank">"/sofia/internal/0412345678@192.168.0.160:5060"</a>.
Is that expected behaviour?<br>
<br>
</div>
</blockquote>
<div>it's not hitting the dialplan, i guess. i suggest you
create prefix 9 for PSTN calls e.g. <br>
<condition field="destination_number"
expression="^9(\d+)$"><br>
<br>
and make the file ../dialplan/default/01_custom.xml is on top
of dialplan/default directory so it will be scanned early.<br>
<br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div bgcolor="#FFFFFF" text="#000000"> 3. I assumed that the
IP address 192.168.0.9 in your example is the address of
your HT503 and not FS. I have thus replaced it with the IP
address from my GXW4104 (192.168.0.160). Is that correct?
<div>
<div><br>
</div>
</div>
</div>
</blockquote>
<div>that is correct. <br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div bgcolor="#FFFFFF" text="#000000">
<div>
<div> <br>
<br>
<br>
<br>
<br>
<br>
On 09/13/2011 11:42 PM, Nandy Dagondon wrote:
<blockquote type="cite">i inserted my answers to your
questions below. for point #3), here's an example how
i configured my FXO port of ht503.<br>
<br>
included in sip_profile/internal:<br>
<include><br>
<gateway name="ht503-fxo"><br>
<param name="username" value="1019"/> <--
it's registered to receive incoming calls<br>
<param name="realm" value="192.168.0.9"/><br>
<param name="sip-port" value="5062"/> <--
port 5060 is set to the FXS port<br>
<param name="password" value="1234"/><br>
<param name="rtp_ip" value="192.168.0.9"/><br>
<param name="dtmf-type" value="rfc2833"/><br>
<param name="expire-seconds" value="600"/><br>
<param name="register" value="false"/><br>
<param name="caller-id-in-from"
value="false"/><br>
</gateway><br>
</include><br>
<br>
included in dialplan/default<br>
<br>
<include><br>
<extension name="ht503-fxo-local"><br>
<condition field="${toll_allow}"
expression="local"/><br>
<condition field="destination_number"
expression="^9([2-9]\d{6})$"><br>
<action application="set"
data="effective_caller_id_number=<a href="tel:0321234567" value="+63321234567" target="_blank">0321234567</a>"/><br>
<action application="set"
data="effective_caller_id_name=ThisIsMy Company"/><br>
<action application="set"
data="ignore_early_media=ring_ready"/> <br>
<action application="set"
data="ringback=${us-ring}"/><br>
<action application="bridge"
data="sofia/gateway/ht503-fxo/$<a href="http://1@192.168.0.9:5062" target="_blank">1@192.168.0.9:5062</a>"/><br>
</condition><br>
</extension><br>
</include><br>
<br>
it looks you can create 4 internal gateways for the
every port, fxo-1 to fxo-4, w/ the same realm/rtp_ip
values but setting different sip-port values. then
your bridge app would be:<br>
<br>
<action application="bridge"
data="sofia/gateway/fxo-1/$1|sofia/gateway/fxo-2/$1|sofia/gateway/fxo-3/$1sofia/gateway/fxo-4/$1"/><br>
<br>
if u want to dialout any free port.<br>
<br>
i haven't tested the above. just try it. i hope it
works.<br>
<br>
-nandy<br>
<br>
<br>
<div class="gmail_quote">On Tue, Sep 13, 2011 at 6:10
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: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div> Hi Nandy<br>
<br>
Thanks for your reply. I assume 192.168.0.9 in
your example is the IP address of the GXW4104? <br>
</div>
</blockquote>
<div>yes. <br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div> <br>
</div>
</blockquote>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div bgcolor="#FFFFFF" text="#000000"> Some more
questions<br>
<br>
1. When you say port number, is this something I
should be setting up on the GXW4104 so that it
is listening on those 4 port numbers? If yes,
what would be the setting I am looking for?<br>
</div>
</blockquote>
<div>not for every port. the gateway has a base port
number e.g. 5060 for port#1. add 2 to the
subsequent ports e.g. 5062 for port#2 and so on.
this is pointed out by sergey. <br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div> <br>
2. Does that mean I don't define a new gateway
in FreeSWITCH?<br>
</div>
</blockquote>
<div>it's an option. but defining a gateway is
cleaner.<br>
<br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div> <br>
</div>
</blockquote>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div bgcolor="#FFFFFF" text="#000000"> 3. In your
example, you said the bridge data would be <a href="mailto:7654321@192.168.0.9:5063" target="_blank">7654321@192.168.0.9:5063</a>.
What would the whole line look like in the
dialplan?<b><br>
<br>
<action application="bridge" data="<a href="mailto:sofia/gateway/7654321@192.168.0.9:5063" target="_blank">sofia/gateway/7654321@192.168.0.9:5063</a></b><b>"/></b><br>
<br>
Still very confused :-)<br>
<br>
Thanks
<div>
<div><br>
<br>
On 09/13/2011 03:45 PM, Nandy Dagondon
wrote:
<blockquote type="cite">hi,<br>
<br>
if GWX4104 is in your local network, use
the internal profile for the gateway.
register your FXO accounts to receive
incoming calls (i think you did this
already). <br>
<br>
to dialout the ports, specify the port
number 5060~5063 assuming Port1 starts at
5060. to dialout via port4, the bridge
data should look like:<br>
<br>
<a href="http://7654321@192.168.0.9:5063" target="_blank">7654321@192.168.0.9:5063</a><br>
<br>
hope it helps.<br>
<br>
-nandy<br>
<br>
<br>
<div class="gmail_quote">On Tue, Sep 13,
2011 at 2:49 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: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div bgcolor="#FFFFFF" text="#000000">
Hi<br>
<br>
I have recently bought a Grandstream
GXW4104 (4 FXO ports) and need some
help setting up a gateway to call
out using the GXW4104. I am really
out of my depth here and may be
looking at this the wrong way so
please bear with me.<br>
<br>
I followed the advice on this
website <a href="http://www.timhunt.net/wiki/FreeSwitch:GXW4104" target="_blank">"http://www.timhunt.net/wiki/FreeSwitch:GXW4104"</a>
and incoming calls from a PSTN line
are working great. Now I need to
setup a dialplan so that outgoing
calls are routed through the same
PSTN line on the GXW4104. I will
eventually have 4 PSTN lines with a
dialplan to use the first available
line (if that is possible).<br>
<br>
According to the FreeSWITCH 1.0.6
book (and many online posts) I need
to create a gateway and a dialplan
but all the gateway examples are for
SIP accounts. <br>
<br>
So, the gateway definition seems to
need a username and password but the
GXW4104 does not have that
capability. I found this gateway
definition in the
freeswitch.xml.fsxml file but am not
sure how many of these variables are
required.<br>
<br>
<gateways><br>
<gateway name="<a href="http://example.com" target="_blank">example.com</a>"><br>
<param name="username"
value="joeuser"/><br>
<param name="password"
value="password"/><br>
<param name="from-user"
value="joeuser"/><br>
<param name="from-domain"
value="<a href="http://example.com" target="_blank">example.com</a>"/><br>
<param
name="expire-seconds"
value="600"/><br>
<param name="register"
value="false"/><br>
<param name="retry-seconds"
value="30"/><br>
<param name="extension"
value="5000"/><br>
<param name="context"
value="public"/><br>
</gateway><br>
</gateways><br>
<br>
<br>
If I define a gateway called
"gxw4104", then this is what I think
a simple dialplan should look like
but I'm not sure of the gateway
details in the "bridge" section of
the definition.<br>
<br>
<extension name="gxw4104_out"><br>
<condition
field="destination_number"
expression="^(\d{10})$"> <br>
<b><action
application="bridge"
data="sofia/gateway/gxw4104/........"/>
(what should this be???)</b><br>
</condition><br>
</extension><br>
<br>
Am I moving in the right direction
and can someone fill in the blanks
for me please<br>
<br>
Thanks in advance!<br>
<br>
<br>
</div>
<br>
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