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chris at fowler.cc Guest
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Posted: Fri Oct 16, 2009 10:13 pm Post subject: [Freeswitch-users] NAT problems migrating from Version 1.0.t |
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Hi,
We've been using 13168M in production for some time now (works great).
I want to get us onto the latest build but am having problems getting
NAT to work.
Phones can register; can dial 9999 test #, but after 100 seconds the
call is disconnected with error:
2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
sofia/internal/1001@myhost.mydomain.com [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]
I took the standard internal.xml and vars.xml files from the new build
and made the following modifications - which worked previously:
modify conf/vars.xml and update
<X-PRE-PROCESS cmd="set" data="domain=myhost.mydomain.com"/>
<X-PRE-PROCESS cmd="set" data="bind_server_ip=1.2.3.4"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/>
Modify conf/sip_profiles/internal.xml
<param name="aggressive-nat-detection" value="true"/> <param
name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param
name="ext-sip-ip" value="$${external_sip_ip}"/> <param
name="NDLB-received-in-nat-reg-contact" value="true"/> <param
name="NDLB-force-rport" value="true"/> <param
name="NDLB-broken-auth-hash" value="true"/>
The big difference I note is that on PRODUCTION (which works) "sofia
status profile internal" yields:
URL sip:mod_sofia@1.2.3.4:5060
BIND-URL sip:mod_sofia@1.2.3.4:5060;maddr=10.250.35.224
But on Test I see:
URL sip:mod_sofia@10.250.66.210:5060
BIND-URL sip:mod_sofia@10.250.66.210:5060
Any ideas?
Thanks, Chris.
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chris at fowler.cc Guest
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Posted: Sun Oct 18, 2009 10:25 pm Post subject: [Freeswitch-users] NAT problems migrating from Version 1.0.t |
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I've tried all sorts of debug and parameter changes over the weekend,
but still can't figure out the correct solution.
If I disable timers on the sip profile then all works fine.
<param name="enable-timer" value="false"/>
But that seems like a hack; not a correct solution. With the build
13168M (which is pre the new NAT functionality) everything worked fine.
The SIP trace shows the phones and FreeSWITCH happily exchanging NOTIFY
and "200 OK" messages. Audio's working - just calls timeout after 100
seconds with RECOVERY_ON_TIMER_EXPIRE. Is enforcement of this timer new
functionality - and really just exposing a problem I've always had
before?
The config is (50 Polycom Phones -> NAT -> Internet -> Amazon EC2)
I would really appreciate some pointers on what to look for; additional
trace that might reveal something.
Thanks, Chris.
On Fri, 16 Oct 2009 20:06:52 -0700, "Chris Fowler" <chris@fowler.cc>
said:
Quote: | Hi,
We've been using 13168M in production for some time now (works great).
I want to get us onto the latest build but am having problems getting
NAT to work.
Phones can register; can dial 9999 test #, but after 100 seconds the
call is disconnected with error:
2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
sofia/internal/1001@myhost.mydomain.com [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]
I took the standard internal.xml and vars.xml files from the new build
and made the following modifications - which worked previously:
modify conf/vars.xml and update
<X-PRE-PROCESS cmd="set" data="domain=myhost.mydomain.com"/>
<X-PRE-PROCESS cmd="set" data="bind_server_ip=1.2.3.4"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/>
Modify conf/sip_profiles/internal.xml
<param name="aggressive-nat-detection" value="true"/> <param
name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param
name="ext-sip-ip" value="$${external_sip_ip}"/> <param
name="NDLB-received-in-nat-reg-contact" value="true"/> <param
name="NDLB-force-rport" value="true"/> <param
name="NDLB-broken-auth-hash" value="true"/>
The big difference I note is that on PRODUCTION (which works) "sofia
status profile internal" yields:
URL sip:mod_sofia@1.2.3.4:5060
BIND-URL sip:mod_sofia@1.2.3.4:5060;maddr=10.250.35.224
But on Test I see:
URL sip:mod_sofia@10.250.66.210:5060
BIND-URL sip:mod_sofia@10.250.66.210:5060
Any ideas?
Thanks, 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
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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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mike at jerris.com Guest
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Posted: Mon Oct 19, 2009 8:19 am Post subject: [Freeswitch-users] NAT problems migrating from Version 1.0.t |
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we added some params for new better automatic nat handling, grep the
new defailt configs for localnet and you will find what you are missing.
Mike
On Oct 18, 2009, at 11:14 PM, Chris Fowler wrote:
Quote: | I've tried all sorts of debug and parameter changes over the weekend,
but still can't figure out the correct solution.
If I disable timers on the sip profile then all works fine.
<param name="enable-timer" value="false"/>
But that seems like a hack; not a correct solution. With the build
13168M (which is pre the new NAT functionality) everything worked
fine.
The SIP trace shows the phones and FreeSWITCH happily exchanging
NOTIFY
and "200 OK" messages. Audio's working - just calls timeout after 100
seconds with RECOVERY_ON_TIMER_EXPIRE. Is enforcement of this timer
new
functionality - and really just exposing a problem I've always had
before?
The config is (50 Polycom Phones -> NAT -> Internet -> Amazon EC2)
I would really appreciate some pointers on what to look for;
additional
trace that might reveal something.
Thanks, Chris.
On Fri, 16 Oct 2009 20:06:52 -0700, "Chris Fowler" <chris@fowler.cc>
said:
Quote: | Hi,
We've been using 13168M in production for some time now (works
great).
I want to get us onto the latest build but am having problems getting
NAT to work.
Phones can register; can dial 9999 test #, but after 100 seconds the
call is disconnected with error:
2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
sofia/internal/1001@myhost.mydomain.com [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]
I took the standard internal.xml and vars.xml files from the new
build
and made the following modifications - which worked previously:
modify conf/vars.xml and update
<X-PRE-PROCESS cmd="set" data="domain=myhost.mydomain.com"/>
<X-PRE-PROCESS cmd="set" data="bind_server_ip=1.2.3.4"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/>
Modify conf/sip_profiles/internal.xml
<param name="aggressive-nat-detection" value="true"/> <param
name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param
name="ext-sip-ip" value="$${external_sip_ip}"/> <param
name="NDLB-received-in-nat-reg-contact" value="true"/> <param
name="NDLB-force-rport" value="true"/> <param
name="NDLB-broken-auth-hash" value="true"/>
The big difference I note is that on PRODUCTION (which works) "sofia
status profile internal" yields:
URL sip:mod_sofia@1.2.3.4:5060
BIND-URL
sip:mod_sofia@1.2.3.4:5060;maddr=10.250.35.224
But on Test I see:
URL sip:mod_sofia@10.250.66.210:5060
BIND-URL sip:mod_sofia@10.250.66.210:5060
Any ideas?
Thanks, Chris.
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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