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@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@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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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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@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@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@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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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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@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@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@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@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@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@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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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3) <param name="disable-rtp-auto-adjust" value="true"/> is not really required at least for my working setup behind the NAT router.

ok, this param is originally added for another problem http://jira.freeswitch.org/browse/MODENDP-198 . But I think it might be useful for this.

Hi Chris. thanks for the reply. Here're my answers. On Thu, Jun 25, 2009 at 9:02 PM, Chris Chen <chris.chen2004@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@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@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@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@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@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@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@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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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I have been testing dingaling all day... I added switch_nat routines to poke holes in the nat if needed if you're behind upnp or nat-pmp. /b On Jun 25, 2009, at 9:30 PM, Chris Chen wrote:

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

Hi Chris, any thoughts? Thanks, -Jingwei On Fri, Jun 26, 2009 at 11:34 AM, Jingwei Yang <jingwei.yang@gmail.com>wrote:

Hi Chris, here's the one that confuses me. As far as I understand, the extension 888 defined in public.xml is for picking up incoming calls. It should have no influence on outgoing calls, right? If not, what is to write to fit my case? (originate dingaling/gmail.com/userAAA@gmail.com&bridge(dingaling/ gmail.com/userBBB@gmail.com), both userAAA and userBBB can be internal or external).

Anyway, I've defined a extension 888 in public.xml. Frankly speaking, I'm not quite sure what to include. So I make it very simple.

<extension name="gtalk"> <condition field="destination_number" expression="^(888)$"> <action application="voicemail" data="default $${domain} 888"/> </condition> </extension>

Here are three relative parameters in client.xml:

<param name="rtp-ip" value="192.168.1.100"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <!--<param name="disable-rtp-auto-adjust" value="true"/>-->

Still, I got no echo for internal Ip calls. Please let me know where goes wrong.

Thanks, -Jingwei

On Fri, Jun 26, 2009 at 10:30 AM, Chris Chen <chris.chen2004@gmail.com>wrote:

...

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

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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>wrote:

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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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>wrote:

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>wrote:

...

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>wrote:

...

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.

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