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[Freeswitch-users] mod_dingaling no audio


 
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chris.chen2004 at gmai...
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PostPosted: Mon Jul 13, 2009 6:32 am    Post subject: [Freeswitch-users] mod_dingaling no audio Reply with quote

Jingwei, can you show your console log when somebody is calling you from gtalk client? Will it really hit 888 in your dialplan?
Thanks,
Chris

On Mon, Jul 13, 2009 at 5:27 AM, Jingwei Yang <jingwei.yang@gmail.com (jingwei.yang@gmail.com)> wrote:
Quote:
Hi Chris, sorry for the late reply. Have been quite busy last few days.

I had shifted 888 from default.xml to public.xml and the dialplan is simply having an echo action now. I've turned on dl_debug but unfortunately didn't find anything useful. Logs are attached for your reference.

I don't think there's something wrong with the dialplan as two external parties can talk to each other perfectly (with ext-rtp-ip uncommented, at this time my ip was interpreted to be an external one). With ext-rtp-ip commented, I can hear the echo and I saw my ip was translated into an internal one (at this time, external party's audio failed).

I tried the method on this wiki page as well: http://wiki.freeswitch.org/wiki/NAT_Traversal (the last FreeSwitch behind NAT portion) but still no luck. Please kindly let me know what other configs I should change.

Thanks,
-Jingwei

On Mon, Jun 29, 2009 at 6:46 PM, Chris Chen <chris.chen2004@gmail.com (chris.chen2004@gmail.com)> wrote:

Quote:
Jingwei, I don't know if you have the 888 defined in default.xml?  also you have to define $${domain}.
please do " dl_debug on" from fs_cli, and watch the console logs and see what's going on when you try calling from external. Most likely your dialplan is not correctly defined.

Chris

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jingwei.yang at gmail.com
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PostPosted: Mon Jul 13, 2009 10:35 pm    Post subject: [Freeswitch-users] mod_dingaling no audio Reply with quote

Hi Chris, I've attached the console logs for your reference. It really hits 888 in the dialplan and the external call can hear the echo without any problem.

One thing attracts me is how the ip addresses are translated. Here's the working external example:

<ses:candidate address="192.168.2.150" port="1726" name="rtp" username="k+TZoLgUQB02PV/C" password="pQ3hz/DzarYnnLhJ" preference="1" protocol="udp" generation="0" network="0" type="local"></ses:candidate>        (external party's local addr)

<ses:candidate address="123.118.149.94" port="45538" name="rtp" username="C5lc0jkpxDBWVRn6" password="u2lw7FbrzmaJJSHq" preference="0.9" protocol="udp" generation="0" network="0" type="stun"></ses:candidate>        (external party's global addr)

<ses:candidate name="rtp" address="59.189.194.244" port="17112" username="PQ0Fmus5M3udzy3G" password="PQ0Fmus5M3udzy3G" preference="1.0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>        (server's global addr)

Here's the non-working internal exmaple:

<candidate name="rtp" address="192.168.1.101" port="2202" preference="1" username="y6P1CToBaRZpybAl" protocol="udp" generation="0" password="tcrX5Wj426qTa1Cx" type="local" network="0"></candidate>        (internal party's local addr)

<candidate name="rtp" address="59.189.194.244" port="2203" preference="0.9" username="UMuUvr9o66D2myqb" protocol="udp" generation="0" password="vvq4+GUsOYV/On5g" type="stun" network="0"></candidate>        (internal party's global addr)

<ses:candidate name="rtp" address="59.189.194.244" port="30578" username="MeymWgoZwasUm32h" password="MeymWgoZwasUm32h" preference="1.0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate>        (server's global addr)

<candidate name="rtp" address="192.168.1.101" port="2202" preference="1" username="y6P1CToBaRZpybAl" protocol="udp" generation="0" password="tcrX5Wj426qTa1Cx" type="local" network="0"></candidate>        (internal party's local addr, again!)
    <candidate name="rtp" address="59.189.194.244" port="2203" preference="0.9" username="UMuUvr9o66D2myqb" protocol="udp" generation="0" password="vvq4+GUsOYV/On5g" type="stun" network="0"></candidate>        (internal party's global addr)

<candidate name="rtp" address="209.85.201.126" port="19295" preference="0.5" username="91jrUtL10nk9VcrZ" protocol="udp" generation="0" password="9s8fNrseavVslvyZ" type="relay" network="0"></candidate>        (google's global addr)

<candidate name="rtp" address="209.85.201.126" port="19294" preference="0.5" username="91jrUtL10nk9VcrZ" protocol="tcp" generation="0" password="9s8fNrseavVslvyZ" type="relay" network="0"></candidate>        (google's global addr)

<candidate name="rtp" address="192.168.1.101" port="2205" preference="0.8" username="g3/qPdtEdo02zm8Y" protocol="tcp" generation="0" password="WYRbJIk5vcToxsqT" type="local" network="0"></candidate>        (internal party's local addr, 3rd time!)

<candidate name="rtp" address="209.85.201.126" port="443" preference="0.5" username="91jrUtL10nk9VcrZ" protocol="ssltcp" generation="0" password="9s8fNrseavVslvyZ" type="relay" network="0"></candidate>        (google's global addr)


And finally, when the call was hung up, the internal one showed an error like this:

<iq to="yyyyyy@gmail.com/talk930C641A" id="308" type="error" from="xxxxxx@gmail.com/Talk.v1046D90E88C">
  <ses:session type="terminate" id="186547325" initiator="xxxxxx@gmail.com/Talk.v1046D90E88C" xmlns:ses="http://www.google.com/session"></ses:session>
  <error type="modify">
    <sta:bad-request xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:bad-request>
    <sta:text xml:lang="en" xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas">unknown session</sta:text>
  </error>
</iq>

Regards,
-Jingwei

On Mon, Jul 13, 2009 at 7:29 PM, Chris Chen <chris.chen2004@gmail.com (chris.chen2004@gmail.com)> wrote:
Quote:
Jingwei, can you show your console log when somebody is calling you from gtalk client? Will it really hit 888 in your dialplan?
Thanks,
Chris


On Mon, Jul 13, 2009 at 5:27 AM, Jingwei Yang <jingwei.yang@gmail.com (jingwei.yang@gmail.com)> wrote:
Quote:
Hi Chris, sorry for the late reply. Have been quite busy last few days.

I had shifted 888 from default.xml to public.xml and the dialplan is simply having an echo action now. I've turned on dl_debug but unfortunately didn't find anything useful. Logs are attached for your reference.

I don't think there's something wrong with the dialplan as two external parties can talk to each other perfectly (with ext-rtp-ip uncommented, at this time my ip was interpreted to be an external one). With ext-rtp-ip commented, I can hear the echo and I saw my ip was translated into an internal one (at this time, external party's audio failed).

I tried the method on this wiki page as well: http://wiki.freeswitch.org/wiki/NAT_Traversal (the last FreeSwitch behind NAT portion) but still no luck. Please kindly let me know what other configs I should change.

Thanks,
-Jingwei

On Mon, Jun 29, 2009 at 6:46 PM, Chris Chen <chris.chen2004@gmail.com (chris.chen2004@gmail.com)> wrote:

Quote:
Jingwei, I don't know if you have the 888 defined in default.xml?  also you have to define $${domain}.
please do " dl_debug on" from fs_cli, and watch the console logs and see what's going on when you try calling from external. Most likely your dialplan is not correctly defined.

Chris

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