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woodydickson at gmail.com Guest
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Posted: Fri Feb 13, 2009 10:43 pm Post subject: [Freeswitch-users] No-media problem with opensips-freeswitch |
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Hi,
I tried to configure opensips as sip proxy and sip registrars and
freeswitch as B2BUA. Everything works until I start to connect sip
clients that are behind ADSL.
Both freeswitch and opensips are on public IP and I am using external
profile as well.
Does anyone have experience in setting up opensips and freeswitch
together and can share the configuration with me?
Thank you very much in advance for any help.
Regards,
Woody
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brian at freeswitch.org Guest
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Posted: Fri Feb 13, 2009 10:56 pm Post subject: [Freeswitch-users] No-media problem with opensips-freeswitch |
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You have let the names of the profiles confuse you. Chances are
you're trying to hair pin the calls out and back into the same nat.
That usually doesn't work. You will need to give me more details
about your setup.
/b
On Feb 13, 2009, at 9:41 PM, Woody Dickson wrote:
Quote: | Hi,
I tried to configure opensips as sip proxy and sip registrars and
freeswitch as B2BUA. Everything works until I start to connect sip
clients that are behind ADSL.
Both freeswitch and opensips are on public IP and I am using external
profile as well.
Does anyone have experience in setting up opensips and freeswitch
together and can share the configuration with me?
Thank you very much in advance for any help.
Regards,
Woody
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woodydickson at gmail.com Guest
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Posted: Sat Feb 14, 2009 9:22 am Post subject: [Freeswitch-users] No-media problem with opensips-freeswitch |
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Hi
My external.xml is just the default configuration:
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="$${external_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
In my opensips.cfg, all the nated traffic is sent to the
external_sip_ip and external_rtp_port.
Is there anything I should add or change to enable media for device behind nat?
Regards,
Woody
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