[Freeswitch-users] GXW4104 gateway setup for outgoing calls
ocset
ocset at the800group.com
Fri Sep 16 09:14:13 MSD 2011
Hi Nandy
Thanks for your help so far - unfortunately this is still not working.
Based on your examples, I have created the following two files
1.
../sip_profile/internal/01_custom.xml
<include>
<param name="multiple-registrations" value="true"/>
<param name="accept-blind-reg" value="true"/>
<param name="accept-blind-auth" value="true"/>
<gateway name="gxw4104-fxo">
<param name="username" value="1019"/>
<param name="password" value="1234"/>
<param name="realm" value="192.168.0.160"/>
<param name="sip-port" value="5060"/>
<param name="rtp_ip" value="192.168.0.160"/>
<param name="dtmf-type" value="rfc2833"/>
<param name="expire-seconds" value="600"/>
<param name="register" value="false"/>
<param name="caller-id-in-from" value="false"/>
</gateway>
</include>
2.
../dialplan/default/01_custom.xml
<include>
<extension name="gxw4104-fxo-local">
<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="^(\d{10})$">
<action application="set" data="effective_caller_id_number=5555555555"/>
<action application="set" data="effective_caller_id_name=ThisIsMyCompany"/>
<action application="set" data="ignore_early_media=ring_ready"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="bridge"
data="sofia/gateway/gxw4104-fxo/$1 at 192.168.0.160:5060"/>
</condition>
</extension>
</include>
Command "sofia status" shows the followiing
internal profile
sip:mod_sofia at 192.168.0.23:5060 RUNNING (0)
internal::gxw4104-fxo gateway
sip:1019 at 192.168.0.160 NOREG
external profile
sip:mod_sofia at 192.168.0.23:5080 RUNNING (0)
external::example.com gateway
sip:joeuser at example.com NOREG
internal-ipv6 profile
sip:mod_sofia@[::1]:5060 RUNNING (0)
192.168.0.23 alias
internal ALIASED
Here is the call log when I try do dial out:
2011-09-16 12:57:30.903616 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/1014 at 192.168.0.23 [0a83e803-b3fc-4a46-bc74-9d6786dac8e2]
2011-09-16 12:57:30.933863 [INFO] mod_dialplan_xml.c:418 Processing User
1014->0412345678 in context default
2011-09-16 12:57:30.950359 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/0412345678 at 192.168.0.160:5060
[24111b8d-25f0-4433-a49f-88973730ebfb]
2011-09-16 12:57:40.957391 [NOTICE] sofia.c:4789 Hangup
sofia/internal/0412345678 at 192.168.0.160:5060 [CS_CONSUME_MEDIA]
[NORMAL_TEMPORARY_FAILURE]
2011-09-16 12:57:40.958554 [NOTICE] switch_core_session.c:1182 Session 7
(sofia/internal/0412345678 at 192.168.0.160:5060) Ended
2011-09-16 12:57:40.958554 [NOTICE] switch_core_session.c:1184 Close
Channel sofia/internal/0412345678 at 192.168.0.160:5060 [CS_DESTROY]
2011-09-16 12:57:40.958554 [INFO] mod_dptools.c:2355 Originate Failed.
Cause: NORMAL_TEMPORARY_FAILURE
2011-09-16 12:57:40.958554 [NOTICE] mod_dptools.c:2418 Hangup
sofia/internal/1014 at 192.168.0.23 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2011-09-16 12:57:40.995191 [NOTICE] switch_core_session.c:1182 Session 6
(sofia/internal/1014 at 192.168.0.23) Ended
2011-09-16 12:57:40.995191 [NOTICE] switch_core_session.c:1184 Close
Channel sofia/internal/1014 at 192.168.0.23 [CS_DESTROY]
I have the following questions
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.
2. The resultant log does not show my gateway being used but instead
shows "/sofia/internal/0412345678 at 192.168.0.160:5060". Is that expected
behaviour?
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?
On 09/13/2011 11:42 PM, Nandy Dagondon wrote:
> i inserted my answers to your questions below. for point #3), here's
> an example how i configured my FXO port of ht503.
>
> included in sip_profile/internal:
> <include>
> <gateway name="ht503-fxo">
> <param name="username" value="1019"/> <-- it's registered to receive
> incoming calls
> <param name="realm" value="192.168.0.9"/>
> <param name="sip-port" value="5062"/> <-- port 5060 is set to the FXS port
> <param name="password" value="1234"/>
> <param name="rtp_ip" value="192.168.0.9"/>
> <param name="dtmf-type" value="rfc2833"/>
> <param name="expire-seconds" value="600"/>
> <param name="register" value="false"/>
> <param name="caller-id-in-from" value="false"/>
> </gateway>
> </include>
>
> included in dialplan/default
>
> <include>
> <extension name="ht503-fxo-local">
> <condition field="${toll_allow}" expression="local"/>
> <condition field="destination_number" expression="^9([2-9]\d{6})$">
> <action application="set" data="effective_caller_id_number=0321234567"/>
> <action application="set" data="effective_caller_id_name=ThisIsMy
> Company"/>
> <action application="set" data="ignore_early_media=ring_ready"/>
> <action application="set" data="ringback=${us-ring}"/>
> <action application="bridge"
> data="sofia/gateway/ht503-fxo/$1 at 192.168.0.9:5062
> <http://1@192.168.0.9:5062>"/>
> </condition>
> </extension>
> </include>
>
> 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:
>
> <action application="bridge"
> data="sofia/gateway/fxo-1/$1|sofia/gateway/fxo-2/$1|sofia/gateway/fxo-3/$1sofia/gateway/fxo-4/$1"/>
>
> if u want to dialout any free port.
>
> i haven't tested the above. just try it. i hope it works.
>
> -nandy
>
>
> On Tue, Sep 13, 2011 at 6:10 PM, ocset <ocset at the800group.com
> <mailto:ocset at the800group.com>> wrote:
>
> Hi Nandy
>
> Thanks for your reply. I assume 192.168.0.9 in your example is the
> IP address of the GXW4104?
>
> yes.
>
>
> Some more questions
>
> 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?
>
> 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.
>
>
> 2. Does that mean I don't define a new gateway in FreeSWITCH?
>
> it's an option. but defining a gateway is cleaner.
>
>
> 3. In your example, you said the bridge data would be
> 7654321 at 192.168.0.9:5063 <mailto:7654321 at 192.168.0.9:5063>. What
> would the whole line look like in the dialplan?*
>
> <action application="bridge"
> data="sofia/gateway/7654321 at 192.168.0.9:5063
> <mailto:sofia/gateway/7654321 at 192.168.0.9:5063>**"/>*
>
> Still very confused :-)
>
> Thanks
>
>
> On 09/13/2011 03:45 PM, Nandy Dagondon wrote:
>> hi,
>>
>> 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).
>>
>> 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:
>>
>> 7654321 at 192.168.0.9:5063 <http://7654321@192.168.0.9:5063>
>>
>> hope it helps.
>>
>> -nandy
>>
>>
>> On Tue, Sep 13, 2011 at 2:49 PM, ocset <ocset at the800group.com
>> <mailto:ocset at the800group.com>> wrote:
>>
>> Hi
>>
>> 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.
>>
>> I followed the advice on this website
>> "http://www.timhunt.net/wiki/FreeSwitch:GXW4104"
>> <http://www.timhunt.net/wiki/FreeSwitch:GXW4104> 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).
>>
>> 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.
>>
>> 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.
>>
>> <gateways>
>> <gateway name="example.com <http://example.com>">
>> <param name="username" value="joeuser"/>
>> <param name="password" value="password"/>
>> <param name="from-user" value="joeuser"/>
>> <param name="from-domain" value="example.com
>> <http://example.com>"/>
>> <param name="expire-seconds" value="600"/>
>> <param name="register" value="false"/>
>> <param name="retry-seconds" value="30"/>
>> <param name="extension" value="5000"/>
>> <param name="context" value="public"/>
>> </gateway>
>> </gateways>
>>
>>
>> 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.
>>
>> <extension name="gxw4104_out">
>> <condition field="destination_number" expression="^(\d{10})$">
>> *<action application="bridge"
>> data="sofia/gateway/gxw4104/........"/> (what should
>> this be???)*
>> </condition>
>> </extension>
>>
>> Am I moving in the right direction and can someone fill in
>> the blanks for me please
>>
>> Thanks in advance!
>>
>>
>>
>>
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