[Freeswitch-users] GXW4104 gateway setup for outgoing calls
ocset
ocset at the800group.com
Tue Sep 20 16:06:21 MSD 2011
I fixed the problem.
The guide I followed for setting up the incoming calls suggested this
setting: "Wait for Dial-Tone(Y/N): ch1-4:Y;"
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.
Thanks for all your help.
On 09/16/2011 03:35 PM, Nandy Dagondon wrote:
> pls see my comments below. you're welcome.
> -nandy
>
> On Fri, Sep 16, 2011 at 1:14 PM, ocset <ocset at the800group.com
> <mailto:ocset at the800group.com>> wrote:
>
> 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"
> <mailto: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
> <mailto:sip:mod_sofia at 192.168.0.23:5060> RUNNING (0)
> internal::gxw4104-fxo gateway sip:1019 at 192.168.0.160
> <mailto:sip:1019 at 192.168.0.160> NOREG
> external profile
> sip:mod_sofia at 192.168.0.23:5080
> <mailto:sip:mod_sofia at 192.168.0.23:5080> RUNNING (0)
> external::example.com <http://example.com> gateway
> sip:joeuser at example.com
> <mailto: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 <tel:2011-09-16> 12:57:30.903616 [NOTICE]
> switch_channel.c:669 New Channel sofia/internal/1014 at 192.168.0.23
> <mailto:sofia/internal/1014 at 192.168.0.23>
> [0a83e803-b3fc-4a46-bc74-9d6786dac8e2]
> 2011-09-16 <tel:2011-09-16> 12:57:30.933863 [INFO]
> mod_dialplan_xml.c:418 Processing User 1014->0412345678 in context
> default
> 2011-09-16 <tel:2011-09-16> 12:57:30.950359 [NOTICE]
> switch_channel.c:669 New Channel
> sofia/internal/0412345678 at 192.168.0.160:5060
> <mailto:sofia/internal/0412345678 at 192.168.0.160:5060>
> [24111b8d-25f0-4433-a49f-88973730ebfb]
> 2011-09-16 <tel:2011-09-16> 12:57:40.957391 [NOTICE] sofia.c:4789
> Hangup sofia/internal/0412345678 at 192.168.0.160:5060
> <mailto:sofia/internal/0412345678 at 192.168.0.160:5060>
> [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> 2011-09-16 <tel:2011-09-16> 12:57:40.958554 [NOTICE]
> switch_core_session.c:1182 Session 7
> (sofia/internal/0412345678 at 192.168.0.160:5060
> <mailto:sofia/internal/0412345678 at 192.168.0.160:5060>) Ended
> 2011-09-16 <tel:2011-09-16> 12:57:40.958554 [NOTICE]
> switch_core_session.c:1184 Close Channel
> sofia/internal/0412345678 at 192.168.0.160:5060
> <mailto:sofia/internal/0412345678 at 192.168.0.160:5060> [CS_DESTROY]
> 2011-09-16 <tel:2011-09-16> 12:57:40.958554 [INFO]
> mod_dptools.c:2355 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE
> 2011-09-16 <tel:2011-09-16> 12:57:40.958554 [NOTICE]
> mod_dptools.c:2418 Hangup sofia/internal/1014 at 192.168.0.23
> <mailto:sofia/internal/1014 at 192.168.0.23> [CS_EXECUTE]
> [NORMAL_TEMPORARY_FAILURE]
> 2011-09-16 <tel:2011-09-16> 12:57:40.995191 [NOTICE]
> switch_core_session.c:1182 Session 6
> (sofia/internal/1014 at 192.168.0.23
> <mailto:sofia/internal/1014 at 192.168.0.23>) Ended
> 2011-09-16 <tel:2011-09-16> 12:57:40.995191 [NOTICE]
> switch_core_session.c:1184 Close Channel
> sofia/internal/1014 at 192.168.0.23
> <mailto: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.
>
> 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.
>
> actually, the user and password entries can be deleted because FS is
> not registering to the FXO gateway.
>
> 2. The resultant log does not show my gateway being used but
> instead shows "/sofia/internal/0412345678 at 192.168.0.160:5060"
> <mailto:/sofia/internal/0412345678 at 192.168.0.160:5060>. Is that
> expected behaviour?
>
> it's not hitting the dialplan, i guess. i suggest you create prefix 9
> for PSTN calls e.g.
> <condition field="destination_number" expression="^9(\d+)$">
>
> and make the file ../dialplan/default/01_custom.xml is on top of
> dialplan/default directory so it will be scanned early.
>
> 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?
>
> that is 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 <tel: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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>>>
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