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brian at freeswitch.org Guest
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Posted: Fri Sep 05, 2008 8:15 pm Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
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Did you happen to
ec2-authorize default -P udp -p 16384-32768
/b
On Sep 5, 2008, at 7:29 PM, Damon Brown wrote:
Quote: | 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
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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
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damon at technicate.com Guest
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Posted: Fri Sep 05, 2008 8:48 pm Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
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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:
Quote: | 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
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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brian at freeswitch.org Guest
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Posted: Fri Sep 05, 2008 8:58 pm Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
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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:
Quote: | 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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damon at technicate.com Guest
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Posted: Sat Sep 06, 2008 12:12 am Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
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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:
Quote: | 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
_______________________________________________
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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diego.viola at gmail.com Guest
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Posted: Sat Sep 06, 2008 12:22 am Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
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in vars.xml
On Sat, Sep 6, 2008 at 1:21 AM, Diego Viola <diego.viola@gmail.com> wrote:
Quote: | 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:
Quote: | 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:
Quote: | 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
_______________________________________________
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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_______________________________________________
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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Back to top |
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diego.viola at gmail.com Guest
|
Posted: Sat Sep 06, 2008 12:22 am Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
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|
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:
Quote: | 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:
Quote: | 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
_______________________________________________
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
|
_______________________________________________
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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brian at freeswitch.org Guest
|
Posted: Sat Sep 06, 2008 12:24 am Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
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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:
Quote: | 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:
Quote: | 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
_______________________________________________
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 |
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damon at technicate.com Guest
|
Posted: Sat Sep 06, 2008 12:29 am Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
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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:
Quote: | 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:
Quote: | Yes, I have all of the valid posts open on my security group
-d
|
Brian West
sip:brian@freeswitch.org
_______________________________________________
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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damon at technicate.com Guest
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Posted: Sat Sep 06, 2008 12:31 am Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
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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:
Quote: | 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:
Quote: | 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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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
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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damon at technicate.com Guest
|
Posted: Mon Sep 08, 2008 9:42 am Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
|
|
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: Quote: | Quote: | 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:
Quote: | 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> (brian@freeswitch.org)
Sent: Friday, September 5, 2008 6:57pm
To: freeswitch-users@lists.freeswitch.org (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:
Quote: | Yes, I have all of the valid posts open on my security group
-d
| Brian West
[url=sip:brian@freeswitch.org]sip:brian@freeswitch.org[/url]
_______________________________________________
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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http://www.freeswitch.org
|
Brian West
[url=sip:brian@freeswitch.org]sip:brian@freeswitch.org[/url]
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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joes.mailing.lists at ... Guest
|
Posted: Mon Sep 08, 2008 10:41 am Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
|
|
Hi,
When I first tried setting up freeswitch on ec2 I attempted to do
this through the quick and dirty wiki:
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
and the ec2 wiki
http://wiki.freeswitch.org/wiki/Amazon_ec2
This didn't work out for me. You also need to take a careful look at
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall
then, for sip, setup a security group with the following ports enabled
(easiest to do with elasticfox
http://developer.amazonwebservices.com/connect/entry.jspa?externalID=609
):
udp 16384:32768
udp 4569
udp 5060
tcp 5060
udp 5080
tcp 5080
tcp 8000
udp 8000
make sure you make the security group, and apply it before you boot
the instance.
after this, the quick and dirty install guid worked just fine for
me. you'll just need to tweak the default dialplan
to your needs.
cheers
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:
Quote: | 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:
Quote: | 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>
|
|
_______________________________________________
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 |
|
Back to top |
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chris at maxpowersoft.com Guest
|
Posted: Mon Sep 08, 2008 10:54 am Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
|
|
Updated the wiki http://wiki.freeswitch.org/wiki/Amazon_ec2 with Novak's notes.
Kind Regards,
Chris
Novak Joe wrote: Quote: | Quote: | Hi,
When I first tried setting up freeswitch on ec2 I attempted to do
this through the quick and dirty wiki:
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
and the ec2 wiki
http://wiki.freeswitch.org/wiki/Amazon_ec2
This didn't work out for me. You also need to take a careful look at
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall
then, for sip, setup a security group with the following ports enabled
(easiest to do with elasticfox
http://developer.amazonwebservices.com/connect/entry.jspa?externalID=609
):
udp 16384:32768
udp 4569
udp 5060
tcp 5060
udp 5080
tcp 5080
tcp 8000
udp 8000
make sure you make the security group, and apply it before you boot
the instance.
after this, the quick and dirty install guid worked just fine for
me. you'll just need to tweak the default dialplan
to your needs.
cheers
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:
Quote: | 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:
Quote: | 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>
| |
_______________________________________________
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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damon at technicate.com Guest
|
Posted: Mon Sep 08, 2008 12:14 pm Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
|
|
I have verified all of my steps based on the wiki. When I make an outbound call (or any call that would affect NAT), I hear the ring for a second than it disappears. The console shows:
2008-09-08 17:06:54 [INFO] mod_sofia.c:1085 sofia_receive_message() Asked to send early media by sofia/internal/1006@75.101.142.208 ([email]sofia/internal/1006@75.101.142.208[/email])
2008-09-08 17:06:54 [INFO] mod_sofia.c:1126 sofia_receive_message() Ring SDP:
v=0
o=FreeSWITCH 1220867834 1220867835 IN IP4 10.251.38.32
s=FreeSWITCH
c=IN IP4 10.251.38.32
t=0 0
a=sendrecv
m=audio 25780 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:30
2008-09-08 17:06:54 [NOTICE] mod_sofia.c:1129 sofia_receive_message() Pre-Answer sofia/internal/1006@75.101.142.208 ([email]sofia/internal/1006@75.101.142.208[/email])!
2008-09-08 17:07:11 [NOTICE] sofia.c:2573 sofia_handle_sip_i_state() Hangup sofia/internal/1006@75.101.142.208 ([email]sofia/internal/1006@75.101.142.208[/email]) [CS_EXECUTE] [ORIGINATOR_CANCEL]
2008-09-08 17:07:11 [NOTICE] switch_ivr_bridge.c:384 audio_bridge_on_exchange_media() Hangup sofia/external/19169621931@sjc-primary.voicepulse.com ([email]sofia/external/19169621931@sjc-primary.voicepulse.com[/email]) [CS_RESET] [NORMAL_CLEARING]
I used the Quick and Dirty Install .... I believe I'm close and just missing something. Any other thoughts?
Thanks all for your help
Damon
Chris Danielson wrote: Quote: | Updated the wiki http://wiki.freeswitch.org/wiki/Amazon_ec2 with Novak's notes.
Kind Regards,
Chris
Novak Joe wrote: Quote: | Quote: | Hi,
When I first tried setting up freeswitch on ec2 I attempted to do
this through the quick and dirty wiki:
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
and the ec2 wiki
http://wiki.freeswitch.org/wiki/Amazon_ec2
This didn't work out for me. You also need to take a careful look at
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall
then, for sip, setup a security group with the following ports enabled
(easiest to do with elasticfox
http://developer.amazonwebservices.com/connect/entry.jspa?externalID=609
):
udp 16384:32768
udp 4569
udp 5060
tcp 5060
udp 5080
tcp 5080
tcp 8000
udp 8000
make sure you make the security group, and apply it before you boot
the instance.
after this, the quick and dirty install guid worked just fine for
me. you'll just need to tweak the default dialplan
to your needs.
cheers
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:
Quote: | 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:
Quote: | 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>
| |
_______________________________________________
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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brian at freeswitch.org Guest
|
Posted: Mon Sep 08, 2008 12:16 pm Post subject: [Freeswitch-users] SIP, NAT and Amazon EC2 |
|
|
You have to set the ext-sip-ip and ext-rtp-ip on the internal profile.
/b
On Sep 8, 2008, at 12:12 PM, Damon Brown wrote:
Quote: | I used the Quick and Dirty Install .... I believe I'm close and just
missing something. Any other thoughts?
|
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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