[Freeswitch-users] mod_dingaling no audio
Chris Chen
chris.chen2004 at gmail.com
Thu Jun 25 19:30:19 PDT 2009
I guess you have the problem here,
in client.xml you have
<param name="context" value="public"/>
but you only define extension 888 in default context,
that's why nobody can reach you from public.
under /usr/local/freeswitch/conf/dialplan
define extension 888 in public.xml to the proper extension you expect, and
check the console log from fs_cli when you do gtalk calling to your gmail
client, you will find out the solution to your issue.
chris
On Thu, Jun 25, 2009 at 10:15 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
> Hi Chris. thanks for the reply. Here're my answers.
>
> On Thu, Jun 25, 2009 at 9:02 PM, Chris Chen <chris.chen2004 at gmail.com>wrote:
>
>> Two questions for you:
>>
>> 1) Do you have extension "888" in your public context?
>
>
> What public context are you saying? I only defined 888.xml in
> /usr/local/freeswitch/conf/directory/default.
>
>
>> 2)Can you put your internal Ip address of FS in rtp-ip instead of
>> $${bind_server_ip} just to make sure it get the right IP?
>
>
> I changed it to the internal Ip, but still no echo.
>
>
>>
>> 3) <param name="disable-rtp-auto-adjust" value="true"/> is not really
>> required at least for my working setup behind the NAT router.
>
>
> Thanks, I've commented it out.
>
>
>>
>>
>>
>> On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>
>>> Hi Chris, thanks for your help. Here's my client.xml
>>>
>>> <include>
>>> <!-- Client Profile (Original mode) -->
>>> <!-- to use this profile take the x- away from the open and close tags
>>> so its <profile> and </profile> -->
>>> <profile type="client">
>>> <param name="name" value="gmail.com"/>
>>> <param name="login" value="userAAA at gmail.com/talk"/>
>>> <param name="password" value="xxxxxxxx"/>
>>> <param name="dialplan" value="XML"/>
>>> <param name="context" value="public"/>
>>> <param name="message" value="Jingle all the way"/>
>>> <param name="rtp-ip" value="$${bind_server_ip}"/>
>>> <!--<param name="ext-sip-ip" value="$${external_sip_ip}"/>-->
>>> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>> <param name="disable-rtp-auto-adjust" value="true"/>
>>> <param name="auto-login" value="true"/>
>>> <!-- SASL "plain" or "md5" -->
>>> <param name="sasl" value="plain"/>
>>> <!-- if the server where the jabber is hosted is not the same as the
>>> one in the jid -->
>>> <param name="server" value="talk.google.com"/>
>>> <!-- Enable TLS or not -->
>>> <param name="tls" value="true"/>
>>> <!-- disable to trade async for more calls -->
>>> <param name="use-rtp-timer" value="true"/>
>>> <!-- default extension (if one cannot be determined) -->
>>> <param name="exten" value="888"/>
>>> <!-- VAD choose one -->
>>> <!-- <param name="vad" value="in"/> -->
>>> <!-- <param name="vad" value="out"/> -->
>>> <param name="vad" value="both"/>
>>> <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>>> <!--<param name="candidate-acl" value="rfc1918.auto"/>-->
>>> </profile>
>>> </include>
>>>
>>>
>>>
>>> On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen <chris.chen2004 at gmail.com>wrote:
>>>
>>>> Please provide your client.xml detail with confidential information
>>>> crossout, I have gtalk client and server working properly behind the NAT.
>>>> I should be able to help you.
>>>>
>>>> Chris
>>>>
>>>>
>>>> On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>>>
>>>>> Thanks seven. External IPs have sound echo this time with ext-rtp-ip
>>>>> uncommented and disable-rtp-auto-adjust=true. However, internal IP has no
>>>>> audio this time no matter what value disable-rtp-auto-adjust is...
>>>>>
>>>>>
>>>>> On Thu, Jun 25, 2009 at 11:24 AM, seven <dujinfang at gmail.com> wrote:
>>>>>
>>>>>> uncomment ext-rtp-ip
>>>>>>
>>>>>> On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote:
>>>>>>
>>>>>> Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and
>>>>>> put "disable-rtp-auto-adjust" inside client.xml. No matter what value this
>>>>>> parameter has (true or false), local IP is able to hear the echo but
>>>>>> external ones still have no audio.
>>>>>>
>>>>>> On Wed, Jun 24, 2009 at 6:01 PM, seven <dujinfang at gmail.com> wrote:
>>>>>>
>>>>>>> search wiki from sth. like disable_rtp_autoajust , I don't remember
>>>>>>> the exact var.
>>>>>>>
>>>>>>> On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote:
>>>>>>>
>>>>>>> Hi Guys,
>>>>>>>
>>>>>>> Here's my situation:
>>>>>>>
>>>>>>> The freeswitch server and my machine are behind the same LAN. If I
>>>>>>> commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by
>>>>>>> *originate dingaling/gmail.com/userAAA at gmail.com &echo*).
>>>>>>>
>>>>>>> However, external calls have no sound at all no matter whether this
>>>>>>> param is commented out or not.
>>>>>>>
>>>>>>> Please kindly let me know what other params to set to resolve this
>>>>>>> issue.
>>>>>>>
>>>>>>> Thanks,
>>>>>>> -Jingwei
>>>>>>> _______________________________________________
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>>>>>
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>>>>
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>>>
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>>
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>
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