[asterisk-users] SIP Ports (1000 to 2000 works)

Discussion:

Al Bochter

2006-11-13 20:47:40 UTC

I was reading the posts and someone said about the default 1000 to 2000
I see in the .conf the default is 10000 to 20000

I found a service that gives inbound DID's in the firewall 5060 and
10000 - 20000 is setup
no workie on the DID

But when I set 5060 , 10000 - 20000 and (Unblocked) 1000 - 2000
Now the DID works fine.

So you me what the standard is
--
Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

Are you outside of the US?
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Email for information: ***@bochterservices.com

(Cellular) 1-712-432-5401

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WebSite: http://www.freeworlddialup.com/

BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email

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http://www.bochterservices.com/?j=store&t=email_security

GOLD PLATING SERVICES
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Vicky

2006-11-13 20:56:43 UTC

Permalink

actually 10000-20000 are rtp ports used by asterisk .. its not really
compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check
ur rtp.conf what range its using and open that in firewall . Default with
asterisk is 10000-20000 unless changed .

Post by Al Bochter
I was reading the posts and someone said about the default 1000 to 2000
I see in the .conf the default is 10000 to 20000
I found a service that gives inbound DID's in the firewall 5060 and
10000 - 20000 is setup
no workie on the DID
But when I set 5060 , 10000 - 20000 and (Unblocked) 1000 - 2000
Now the DID works fine.
So you me what the standard is
--
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.
(Cellular) 1-712-432-5401
(Voip PBX) Free World DialUp: 780-217 EXT: 250
WebSite: http://www.freeworlddialup.com/
BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email
For new and used security items
http://www.bochterservices.com/?j=store&t=email_security
GOLD PLATING SERVICES
http://www.bochterservices.com/?j=plating&t=email
_______________________________________________
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Al Bochter

2006-11-13 22:39:59 UTC

Permalink

Yes you are right 10000-20000 are rtp ports used by asterisk by default
I have some that do set a custom range in /etc/asterisk/rtp.conf ..

After looking around.. There were not any notes about the 1000 - 2000
port range on there website.
As you know if you don't know what the ports are it no workie!!!!!
And it is not good to DMZ the server.....
----------
Now I have a handytone 386 that is set to

SIP port 5060 and 5062
RTP port 5004 and 5008

You can set Random Ports to use: 1024 to 65535

The handytone will work fine on the LAN.... But if you would moved the
Handytone to the internet it would NOT work do to the firewall..
Using the asterisk defaults
----------
So liked I ask before "So is there any standard ports"

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.

Email for information: ***@bochterservices.com

(Cellular) 1-712-432-5401

(Voip PBX) Free World DialUp: 780-217 EXT: 250
WebSite: http://www.freeworlddialup.com/

BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email

For new and used security items
http://www.bochterservices.com/?j=store&t=email_security

GOLD PLATING SERVICES
http://www.bochterservices.com/?j=plating&t=email

Post by Vicky
actually 10000-20000 are rtp ports used by asterisk .. its not really
compulsary .. you can set a custom range in /etc/asterisk/rtp.conf ..
check ur rtp.conf what range its using and open that in firewall .
Default with asterisk is 10000-20000 unless changed .
I was reading the posts and someone said about the default 1000 to 2000
I see in the .conf the default is 10000 to 20000
I found a service that gives inbound DID's in the firewall 5060 and
10000 - 20000 is setup
no workie on the DID
But when I set 5060 , 10000 - 20000 and (Unblocked) 1000 - 2000
Now the DID works fine.
So you me what the standard is
--
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.
(Cellular) 1-712-432-5401
(Voip PBX) Free World DialUp: 780-217 EXT: 250
WebSite: http://www.freeworlddialup.com/
BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email
<http://www.bochterservices.com/?j=gold&t=email>
For new and used security items
http://www.bochterservices.com/?j=store&t=email_security
<http://www.bochterservices.com/?j=store&t=email_security>
GOLD PLATING SERVICES
http://www.bochterservices.com/?j=plating&t=email
<http://www.bochterservices.com/?j=plating&t=email>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com
<http://Easynews.com> --
asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
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------------------------------------------------------------------------
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Inbound (clean). Database: 0648-1, 11/13/2006 - 11/13/2006 4:03:01 PM

Vicky

2006-11-13 22:52:05 UTC

Permalink

FRom voip-info.org

# SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well
iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT
# IAX2- the IAX protocol
iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT
# IAX - most have switched to IAX v2, or ought to
iptables -A INPUT -p udp -m udp --dport 5036 -j ACCEPT
# RTP - the media stream
iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT
# MGCP - if you use media gateway control protocol in your configuration
iptables -A INPUT -p udp -m udp --dport 2727 -j ACCEPT

Open all above ports and you should be good to go . Maybe you are recieving
calls over iax and u havent opened iax2 port 4569 .. Anyway my server has
all above ports opened and i have zero problems :) .

Post by Al Bochter
Yes you are right 10000-20000 are rtp ports used by asterisk by default
I have some that do set a custom range in /etc/asterisk/rtp.conf ..
After looking around.. There were not any notes about the 1000 - 2000 port
range on there website.
As you know if you don't know what the ports are it no workie!!!!!
And it is not good to DMZ the server.....
----------
Now I have a handytone 386 that is set to
SIP port 5060 and 5062
RTP port 5004 and 5008
You can set Random Ports to use: 1024 to 65535
The handytone will work fine on the LAN.... But if you would moved the
Handytone to the internet it would NOT work do to the firewall..
Using the asterisk defaults
----------
So liked I ask before "So is there any standard ports"
actually 10000-20000 are rtp ports used by asterisk .. its not really
compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check
ur rtp.conf what range its using and open that in firewall . Default with
asterisk is 10000-20000 unless changed .

------------------------------
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---------------------------------------------------- Inbound (clean). Database: 0648-1, 11/13/2006 - 11/13/2006 4:03:01 PM
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Peter Bowyer

2006-11-13 22:52:29 UTC

Permalink

Both sides have to be willing to negotiate a port. Maybe your
handytone has its own restrictions on RTP ports? As you now know,
Asterisk doesn't care as long as you specify a range in rtp.conf.

1000-2000 must be a typo as ports <1024 are reserved and privileged.

There's no standard - there are several different conventions adopted
by different vendors, though.

http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help.

Peter

--
Peter Bowyer
Email: ***@bowyer.org

Al Bochter

2006-11-14 01:12:00 UTC

Permalink

No 1000 to 2000 is not a typo.
Well let me put some light on this......

If you goto http://www.ipkall.com/
and your firewall is set to 10000 to 20000 you WILL NOT get SIP calls
from http://www.ipkall.com/ DID's

As soon as you OPEN ports 1000 to 2000 to the PBX Server the calls from
http://www.ipkall.com/ will work fine.

You DON'T have to make any changes to /etc/asterisk/rtp.conf

This is what I ran into today

So I guess you are right... It's a free for all on ports. Makes things
harder to do.
I think we need to get a better standard just to make this easier.

// There's no standard - there are several different conventions adopted
// by different vendors, though.

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.

Email for information: ***@bochterservices.com

(Cellular) 1-712-432-5401

(Voip PBX) Free World DialUp: 780-217 EXT: 250
WebSite: http://www.freeworlddialup.com/

BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email

For new and used security items
http://www.bochterservices.com/?j=store&t=email_security

GOLD PLATING SERVICES
http://www.bochterservices.com/?j=plating&t=email

Post by Peter Bowyer

Both sides have to be willing to negotiate a port. Maybe your
handytone has its own restrictions on RTP ports? As you now know,
Asterisk doesn't care as long as you specify a range in rtp.conf.
1000-2000 must be a typo as ports <1024 are reserved and privileged.
There's no standard - there are several different conventions adopted
by different vendors, though.
http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help.
Peter

Vicky

2006-11-14 07:28:11 UTC

Permalink

There is definitely wrong in your setup . I have ipkall setup on my
asterisk and dont have ports 1000-2000 open ( only 10000-20000,5060,4569
open ) . and incoming calls word fine for me .

Post by Al Bochter
No 1000 to 2000 is not a typo.
Well let me put some light on this......
If you goto http://www.ipkall.com/
and your firewall is set to 10000 to 20000 you WILL NOT get SIP calls
from http://www.ipkall.com/ DID's
As soon as you OPEN ports 1000 to 2000 to the PBX Server the calls from
http://www.ipkall.com/ will work fine.
You DON'T have to make any changes to /etc/asterisk/rtp.conf
This is what I ran into today
So I guess you are right... It's a free for all on ports. Makes things
harder to do.
I think we need to get a better standard just to make this easier.
// There's no standard - there are several different conventions adopted
// by different vendors, though.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.
(Cellular) 1-712-432-5401
(Voip PBX) Free World DialUp: 780-217 EXT: 250
WebSite: http://www.freeworlddialup.com/
BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email
For new and used security items
http://www.bochterservices.com/?j=store&t=email_security
GOLD PLATING SERVICES
http://www.bochterservices.com/?j=plating&t=email

Post by Peter Bowyer

Both sides have to be willing to negotiate a port. Maybe your
handytone has its own restrictions on RTP ports? As you now know,
Asterisk doesn't care as long as you specify a range in rtp.conf.
1000-2000 must be a typo as ports <1024 are reserved and privileged.
There's no standard - there are several different conventions adopted
by different vendors, though.
http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help.
Peter

_______________________________________________
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asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

Al Bochter

2006-11-14 08:39:27 UTC

Permalink

Where is your DMZ pointed?

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.

Email for information: ***@bochterservices.com

(Cellular) 1-712-432-5401

(Voip PBX) Free World DialUp: 780-217 EXT: 250
WebSite: http://www.freeworlddialup.com/

BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email

For new and used security items
http://www.bochterservices.com/?j=store&t=email_security

GOLD PLATING SERVICES
http://www.bochterservices.com/?j=plating&t=email

Post by Vicky
There is definitely wrong in your setup . I have ipkall setup on my
asterisk and dont have ports 1000-2000 open ( only
10000-20000,5060,4569 open ) . and incoming calls word fine for me .
No 1000 to 2000 is not a typo.
Well let me put some light on this......
If you goto http://www.ipkall.com/
and your firewall is set to 10000 to 20000 you WILL NOT get SIP calls
from http://www.ipkall.com/ DID's
As soon as you OPEN ports 1000 to 2000 to the PBX Server the calls from
http://www.ipkall.com/ will work fine.
You DON'T have to make any changes to /etc/asterisk/rtp.conf
This is what I ran into today
So I guess you are right... It's a free for all on ports. Makes things
harder to do.
I think we need to get a better standard just to make this easier.
// There's no standard - there are several different conventions adopted
// by different vendors, though.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.
(Cellular) 1-712-432-5401
(Voip PBX) Free World DialUp: 780-217 EXT: 250
WebSite: http://www.freeworlddialup.com/
BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email
<http://www.bochterservices.com/?j=gold&t=email>
For new and used security items
http://www.bochterservices.com/?j=store&t=email_security
<http://www.bochterservices.com/?j=store&t=email_security>
GOLD PLATING SERVICES
http://www.bochterservices.com/?j=plating&t=email
<http://www.bochterservices.com/?j=plating&t=email>

Post by Peter Bowyer

Post by Al Bochter
Yes you are right 10000-20000 are rtp ports used by asterisk by

default

Post by Peter Bowyer

Post by Al Bochter
I have some that do set a custom range in /etc/asterisk/rtp.conf ..
After looking around.. There were not any notes about the 1000

- 2000

Post by Peter Bowyer

Post by Al Bochter
port
range on there website.
As you know if you don't know what the ports are it no workie!!!!!
And it is not good to DMZ the server.....
----------
Now I have a handytone 386 that is set to
SIP port 5060 and 5062
RTP port 5004 and 5008
You can set Random Ports to use: 1024 to 65535
The handytone will work fine on the LAN.... But if you would

moved the

Post by Peter Bowyer

Post by Al Bochter
Handytone to the internet it would NOT work do to the firewall..
Using the asterisk defaults
----------
So liked I ask before "So is there any standard ports"

Both sides have to be willing to negotiate a port. Maybe your
handytone has its own restrictions on RTP ports? As you now know,
Asterisk doesn't care as long as you specify a range in rtp.conf.
1000-2000 must be a typo as ports <1024 are reserved and privileged.
There's no standard - there are several different conventions

adopted

Post by Peter Bowyer
by different vendors, though.
http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might

help.

Post by Peter Bowyer
Peter

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com
<http://Easynews.com> --
asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------------------------------------------------
----------------------------------------------------
Inbound (clean). Database: 0648-1, 11/13/2006 - 11/14/2006 2:29:51 AM

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