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jingwei.yang at gmail.com Guest
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Posted: Wed Jun 24, 2009 4:42 am Post subject: [Freeswitch-users] mod_dingaling no audio |
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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 |
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dujinfang at gmail.com Guest
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Posted: Wed Jun 24, 2009 5:01 am Post subject: [Freeswitch-users] mod_dingaling no audio |
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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: |
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jingwei.yang at gmail.com Guest
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Posted: Wed Jun 24, 2009 9:25 pm Post subject: [Freeswitch-users] mod_dingaling no audio |
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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 (dujinfang@gmail.com)> wrote:
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dujinfang at gmail.com Guest
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Posted: Wed Jun 24, 2009 10:26 pm Post subject: [Freeswitch-users] mod_dingaling no audio |
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uncomment ext-rtp-ip
On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote: |
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jingwei.yang at gmail.com Guest
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Posted: Wed Jun 24, 2009 10:42 pm Post subject: [Freeswitch-users] mod_dingaling no audio |
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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 (dujinfang@gmail.com)> wrote:
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chris.chen2004 at gmai... Guest
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Posted: Wed Jun 24, 2009 10:55 pm Post subject: [Freeswitch-users] mod_dingaling no audio |
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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 (jingwei.yang@gmail.com)> wrote:
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jingwei.yang at gmail.com Guest
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Posted: Thu Jun 25, 2009 12:33 am Post subject: [Freeswitch-users] mod_dingaling no audio |
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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="$${external_rtp_ip}"/>
<param name="disable-rtp-auto-adjust" value="true"/>
<param name="auto-login" value="true"/>
<!-- SASL "plain" or "md5" -->
<param name="sasl" value="plain"/>
<!-- if the server where the jabber is hosted is not the same as the one in the jid -->
<param name="server" value="talk.google.com"/>
<!-- Enable TLS or not -->
<param name="tls" value="true"/>
<!-- disable to trade async for more calls -->
<param name="use-rtp-timer" value="true"/>
<!-- default extension (if one cannot be determined) -->
<param name="exten" value="888"/>
<!-- VAD choose one -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<param name="vad" value="both"/>
<!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
<!--<param name="candidate-acl" value="rfc1918.auto"/>-->
</profile>
</include>
On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen <chris.chen2004@gmail.com (chris.chen2004@gmail.com)> wrote:
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chris.chen2004 at gmai... Guest
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Posted: Thu Jun 25, 2009 8:06 am Post subject: [Freeswitch-users] mod_dingaling no audio |
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Two questions for you:
1) Do you have extension "888" in your public context?
2)Can you put your internal Ip address of FS in rtp-ip instead of $${bind_server_ip} just to make sure it get the right IP?
3) <param name="disable-rtp-auto-adjust" value="true"/> is not really required at least for my working setup behind the NAT router.
On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang <jingwei.yang@gmail.com (jingwei.yang@gmail.com)> wrote:
Quote: | 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 (chris.chen2004@gmail.com)> wrote:
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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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dujinfang at gmail.com Guest
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anthony.minessale at g... Guest
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Posted: Thu Jun 25, 2009 10:33 am Post subject: [Freeswitch-users] mod_dingaling no audio |
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if you are behind nat , you will not want to disable auto-adjust that is what the feature is there to help you with.
On Thu, Jun 25, 2009 at 10:16 AM, Seven Du <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
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jingwei.yang at gmail.com Guest
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Posted: Thu Jun 25, 2009 9:17 pm Post subject: [Freeswitch-users] mod_dingaling no audio |
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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 (chris.chen2004@gmail.com)> wrote:
Quote: | 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.
Quote: |
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.
Quote: |
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.
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chris.chen2004 at gmai... Guest
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Posted: Thu Jun 25, 2009 9:32 pm Post subject: [Freeswitch-users] mod_dingaling no audio |
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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@gmail.com (jingwei.yang@gmail.com)> wrote:
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brian at freeswitch.org Guest
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Posted: Thu Jun 25, 2009 9:40 pm Post subject: [Freeswitch-users] mod_dingaling no audio |
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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:
Quote: | 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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_______________________________________________
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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jingwei.yang at gmail.com Guest
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Posted: Thu Jun 25, 2009 10:35 pm Post subject: [Freeswitch-users] mod_dingaling no audio |
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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 (chris.chen2004@gmail.com)> wrote:
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jingwei.yang at gmail.com Guest
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Posted: Mon Jun 29, 2009 4:27 am Post subject: [Freeswitch-users] mod_dingaling no audio |
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Hi Chris, any thoughts?
Thanks,
-Jingwei
On Fri, Jun 26, 2009 at 11:34 AM, Jingwei Yang <jingwei.yang@gmail.com (jingwei.yang@gmail.com)> wrote:
Quote: | 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 (chris.chen2004@gmail.com)> wrote:
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