SIP, NAT and Amazon EC2
Hello All, I have installed freeswitch on a computing cloud (Amazon EC2). My main network configurations are: (FreeSwitch ARI LAN ) <---> (WAN) <------{INTERNET} -------> (WAN) <----> (SIP Soft or Hard Phone LAN) I have pointed all of my clients to the external (5080) sip port .... No matter what I try STUN, Port Forwarding or the "Use Asterisk Method" settings in the directory, I get no audio. My guess would be the RTP side, but at this point I feel like I have done everything I can and still be a FreeSwitch Newb. Can anyone provide any in-site? Seems like a great system ... more directly the idea of no hardware dependency! Thanks in advance Damon
Did you happen to ec2-authorize default -P udp -p 16384-32768 /b On Sep 5, 2008, at 7:29 PM, Damon Brown wrote:
Hello All,
I have installed freeswitch on a computing cloud (Amazon EC2). My main network configurations are:
(FreeSwitch ARI LAN ) <---> (WAN) <------{INTERNET} -------> (WAN) <----> (SIP Soft or Hard Phone LAN)
I have pointed all of my clients to the external (5080) sip port ....
No matter what I try STUN, Port Forwarding or the "Use Asterisk Method" settings in the directory, I get no audio. My guess would be the RTP side, but at this point I feel like I have done everything I can and still be a FreeSwitch Newb. Can anyone provide any in-site?
Seems like a great system ... more directly the idea of no hardware dependency! Thanks in advance
Damon <damon.vcf>_______________________________________________ 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
Brian West sip:brian@freeswitch.org
Yes, I have all of the valid posts open on my security group -d -----Original Message----- From: "Brian West" <brian@freeswitch.org> Sent: Friday, September 5, 2008 6:14pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 Did you happen to ec2-authorize default -P udp -p 16384-32768 /b On Sep 5, 2008, at 7:29 PM, Damon Brown wrote:
Hello All,
I have installed freeswitch on a computing cloud (Amazon EC2). My main network configurations are:
(FreeSwitch ARI LAN ) <---> (WAN) <------{INTERNET} -------> (WAN) <----> (SIP Soft or Hard Phone LAN)
I have pointed all of my clients to the external (5080) sip port ....
No matter what I try STUN, Port Forwarding or the "Use Asterisk Method" settings in the directory, I get no audio. My guess would be the RTP side, but at this point I feel like I have done everything I can and still be a FreeSwitch Newb. Can anyone provide any in-site?
Seems like a great system ... more directly the idea of no hardware dependency! Thanks in advance
Damon <damon.vcf>_______________________________________________ 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
Brian West sip:brian@freeswitch.org _______________________________________________ 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
Ive Tried the following with no success: internal.xml <param name="ext-rtp-ip" value="75.101.142.208"/> <param name="ext-sip-ip" value="75.101.142.208"/> Ive also tried just changing the vars.xml file. I also tried forwarding the incoming rtp connections to my test pc. all with no audio success. I am sure im doing something wrong I jsut cant find it. I found another suggestion on the wiki of creating a "double nat" that listens on 5090. That didnt change anything either, here is that information: <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <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="aggressive-nat-detection" value="true"/> <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="enable-3pcc" value="true"/>--> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="false"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5061 will be used if unspecified)$ <param name="tls-sip-port" value="5081"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for $ <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work w$ <param name="tls-version" value="tlsv1"/> </settings> Maybe someone can see what is going on here. Thanks, Damon -----Original Message----- From: "Brian West" <brian@freeswitch.org> Sent: Friday, September 5, 2008 6:57pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml profile on ec2 duplicate them from the external profile. /b On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
Yes, I have all of the valid posts open on my security group -d
Brian West sip:brian@freeswitch.org _______________________________________________ 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
Try changing these lines and put your ip instead, that worked for me. <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/> Diego On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <damon@technicate.com> wrote:
Ive Tried the following with no success:
internal.xml
<param name="ext-rtp-ip" value="75.101.142.208"/> <param name="ext-sip-ip" value="75.101.142.208"/>
Ive also tried just changing the vars.xml file. I also tried forwarding the incoming rtp connections to my test pc. all with no audio success. I am sure im doing something wrong I jsut cant find it.
I found another suggestion on the wiki of creating a "double nat" that listens on 5090. That didnt change anything either, here is that information:
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${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="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <!--<param name="enable-3pcc" value="true"/>--> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="false"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5061 will be used if unspecified)$ <param name="tls-sip-port" value="5081"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for $ <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work w$ <param name="tls-version" value="tlsv1"/>
</settings>
Maybe someone can see what is going on here.
Thanks, Damon
-----Original Message----- From: "Brian West" <brian@freeswitch.org> Sent: Friday, September 5, 2008 6:57pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml profile on ec2 duplicate them from the external profile.
/b
On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
Yes, I have all of the valid posts open on my security group -d
Brian West sip:brian@freeswitch.org
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in vars.xml On Sat, Sep 6, 2008 at 1:21 AM, Diego Viola <diego.viola@gmail.com> wrote:
Try changing these lines and put your ip instead, that worked for me.
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
Diego
On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <damon@technicate.com> wrote:
Ive Tried the following with no success:
internal.xml
<param name="ext-rtp-ip" value="75.101.142.208"/> <param name="ext-sip-ip" value="75.101.142.208"/>
Ive also tried just changing the vars.xml file. I also tried forwarding the incoming rtp connections to my test pc. all with no audio success. I am sure im doing something wrong I jsut cant find it.
I found another suggestion on the wiki of creating a "double nat" that listens on 5090. That didnt change anything either, here is that information:
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <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="aggressive-nat-detection" value="true"/> <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="enable-3pcc" value="true"/>--> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="false"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5061 will be used if unspecified)$ <param name="tls-sip-port" value="5081"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for $ <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work w$ <param name="tls-version" value="tlsv1"/>
</settings>
Maybe someone can see what is going on here.
Thanks, Damon
-----Original Message----- From: "Brian West" <brian@freeswitch.org> Sent: Friday, September 5, 2008 6:57pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml profile on ec2 duplicate them from the external profile.
/b
On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
Yes, I have all of the valid posts open on my security group -d
Brian West sip:brian@freeswitch.org
_______________________________________________ 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 Diego, I have : <X-PRE-PROCESS cmd="set" data="external_rtp_ip=75.101.142.208"/> <!-- external_sip_ip Used as the public IP address for SDP. Can be an ip address or a string like "stun:stun.server.com" If unspecified, the bind_server_ip value is used. Used by: sofia.conf.xml dingaling.conf.xml --> <X-PRE-PROCESS cmd="set" data="external_sip_ip=75.101.142.208"/> and im still having no luck .... my guess would be some issue with the elastic ip translation? then again im drawing at newbie straws .... im good with the others and nat/rtp have never been an issue ... is here. -----Original Message----- From: "Diego Viola" <diego.viola@gmail.com> Sent: Friday, September 5, 2008 10:21pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 Try changing these lines and put your ip instead, that worked for me. <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/> Diego On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <damon@technicate.com> wrote:
Ive Tried the following with no success:
internal.xml
<param name="ext-rtp-ip" value="75.101.142.208"/> <param name="ext-sip-ip" value="75.101.142.208"/>
Ive also tried just changing the vars.xml file. I also tried forwarding the incoming rtp connections to my test pc. all with no audio success. I am sure im doing something wrong I jsut cant find it.
I found another suggestion on the wiki of creating a "double nat" that listens on 5090. That didnt change anything either, here is that information:
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${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="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <!--<param name="enable-3pcc" value="true"/>--> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="false"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5061 will be used if unspecified)$ <param name="tls-sip-port" value="5081"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for $ <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work w$ <param name="tls-version" value="tlsv1"/>
</settings>
Maybe someone can see what is going on here.
Thanks, Damon
-----Original Message----- From: "Brian West" <brian@freeswitch.org> Sent: Friday, September 5, 2008 6:57pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml profile on ec2 duplicate them from the external profile.
/b
On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
Yes, I have all of the valid posts open on my security group -d
Brian West sip:brian@freeswitch.org
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Let me launch mine in 32bit and see what I can do over the weekend... I had this working without a problem. ;/ /b On Sep 6, 2008, at 12:09 AM, Damon Brown wrote:
Ive Tried the following with no success:
internal.xml
<param name="ext-rtp-ip" value="75.101.142.208"/> <param name="ext-sip-ip" value="75.101.142.208"/>
Ive also tried just changing the vars.xml file. I also tried forwarding the incoming rtp connections to my test pc. all with no audio success. I am sure im doing something wrong I jsut cant find it.
I found another suggestion on the wiki of creating a "double nat" that listens on 5090. That didnt change anything either, here is that information:
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <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="aggressive-nat-detection" value="true"/> <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="enable-3pcc" value="true"/>--> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="false"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5061 will be used if unspecified)$ <param name="tls-sip-port" value="5081"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for $ <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work w$ <param name="tls-version" value="tlsv1"/>
</settings>
Maybe someone can see what is going on here.
Thanks, Damon
-----Original Message----- From: "Brian West" <brian@freeswitch.org> Sent: Friday, September 5, 2008 6:57pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml profile on ec2 duplicate them from the external profile.
/b
On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
Yes, I have all of the valid posts open on my security group -d
Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org
Great thanks ... i look forward to your results. I installed on a deb ARI -----Original Message----- From: "Brian West" <brian@freeswitch.org> Sent: Friday, September 5, 2008 10:23pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2 Let me launch mine in 32bit and see what I can do over the weekend... I had this working without a problem. ;/ /b On Sep 6, 2008, at 12:09 AM, Damon Brown wrote:
Ive Tried the following with no success:
internal.xml
<param name="ext-rtp-ip" value="75.101.142.208"/> <param name="ext-sip-ip" value="75.101.142.208"/>
Ive also tried just changing the vars.xml file. I also tried forwarding the incoming rtp connections to my test pc. all with no audio success. I am sure im doing something wrong I jsut cant find it.
I found another suggestion on the wiki of creating a "double nat" that listens on 5090. That didnt change anything either, here is that information:
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${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="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <!--<param name="enable-3pcc" value="true"/>--> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="false"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5061 will be used if unspecified)$ <param name="tls-sip-port" value="5081"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for $ <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work w$ <param name="tls-version" value="tlsv1"/>
</settings>
Maybe someone can see what is going on here.
Thanks, Damon
-----Original Message----- From: "Brian West" <brian@freeswitch.org> Sent: Friday, September 5, 2008 6:57pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml profile on ec2 duplicate them from the external profile.
/b
On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
Yes, I have all of the valid posts open on my security group -d
Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org _______________________________________________ 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 Brian, Did you have any luck with the EC2 instances. I tried a few things, but I just cant get RTP to run through it. PLease let me know. Best Damon Brian West wrote:
Let me launch mine in 32bit and see what I can do over the weekend... I had this working without a problem. ;/
/b
On Sep 6, 2008, at 12:09 AM, Damon Brown wrote:
Ive Tried the following with no success:
internal.xml
<param name="ext-rtp-ip" value="75.101.142.208"/> <param name="ext-sip-ip" value="75.101.142.208"/>
Ive also tried just changing the vars.xml file. I also tried forwarding the incoming rtp connections to my test pc. all with no audio success. I am sure im doing something wrong I jsut cant find it.
I found another suggestion on the wiki of creating a "double nat" that listens on 5090. That didnt change anything either, here is that information:
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <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="aggressive-nat-detection" value="true"/> <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="enable-3pcc" value="true"/>--> <!-- TLS: disabled by default, set to "true" to enable --> <param name="tls" value="false"/> <!-- additional bind parameters for TLS --> <param name="tls-bind-params" value="transport=tls"/> <!-- Port to listen on for TLS requests. (5061 will be used if unspecified)$ <param name="tls-sip-port" value="5081"/> <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for $ <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work w$ <param name="tls-version" value="tlsv1"/>
</settings>
Maybe someone can see what is going on here.
Thanks, Damon
-----Original Message----- From: "Brian West" <brian@freeswitch.org> Sent: Friday, September 5, 2008 6:57pm To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml profile on ec2 duplicate them from the external profile.
/b
On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
Yes, I have all of the valid posts open on my security group -d
Brian West sip:brian@freeswitch.org
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Brian West sip:brian@freeswitch.org
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participants (3)
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Brian West
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Damon Brown
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Diego Viola