Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Multiple SIP phones behind a Linksys firewa


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
john at quonix.net
Guest





PostPosted: Sat Feb 02, 2008 3:11 pm    Post subject: [asterisk-users] Multiple SIP phones behind a Linksys firewa Reply with quote

I posted an email a few days regarding a problem with hearing the
voicemail greeting on my sip phones.

It turns out to be a phone/stun/linksys issue - not an asterisk issue.
Which brings up a couple of questions....

I always assumed that you can have multiple SIP phones behind a Linksys
firewall/router (WRT54G) all using the same STUN server/port.

But apparently thats not the case. Is it a Linksys bug, a Grandstream bug
in the BudgeTone-100 phone, or am I off base and just doing something
wrong?

I cleary have problems as soon as I try to use a second phone behind the
Linksys - registration issues, cant hear voicemail greeting, etc.,.

My next test was to run multiple STUN servers on the same machine with
different ports. Then, for my multiple SIP phones behind the Linksys, have
each phone use a different stun port.

Any thoughts?

John
Back to top
greg.oliver at cistera...
Guest





PostPosted: Sat Feb 02, 2008 4:15 pm    Post subject: [asterisk-users] Multiple SIP phones behind a Linksys firewa Reply with quote

On Feb 2, 2008, at 2:11 PM, John Von Essen <john at quonix.net> wrote:

Quote:
I posted an email a few days regarding a problem with hearing the
voicemail greeting on my sip phones.

It turns out to be a phone/stun/linksys issue - not an asterisk issue.
Which brings up a couple of questions....

I always assumed that you can have multiple SIP phones behind a
Linksys
firewall/router (WRT54G) all using the same STUN server/port.

But apparently thats not the case. Is it a Linksys bug, a
Grandstream bug
in the BudgeTone-100 phone, or am I off base and just doing something
wrong?

I cleary have problems as soon as I try to use a second phone behind
the
Linksys - registration issues, cant hear voicemail greeting, etc.,.

My next test was to run multiple STUN servers on the same machine with
different ports. Then, for my multiple SIP phones behind the
Linksys, have
each phone use a different stun port.

Any thoughts?

John

I have 3 phones connected to 2 servers behind a 54g running openwrt
with no stun or any special configuration. I am running cisco phones
which do nat well natively.

-greg
Quote:

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
greg.oliver at cistera...
Guest





PostPosted: Sat Feb 02, 2008 6:00 pm    Post subject: [asterisk-users] Multiple SIP phones behind a Linksys firewa Reply with quote

On Feb 2, 2008, at 3:43 PM, john at quonix.net wrote:

Quote:
Greg,

Without STUN how are the phones able to register? I was unable to
get the
Grandstream phones to work at all without STUN.

-John


I have nat on in sip.conf and off on the phones. Works perfect for
7960/1 and 7971. When I get back home, I will login to the asterisk
servers and tell you what IPs the registration requests have in them.
Quote:
----------------------------------------------------
From : Greg Oliver <greg.oliver at cistera.com>
To : Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Subject : Re: [asterisk-users] Multiple SIP phones behind a Linksys
firewall
Date : Sat, 2 Feb 2008 15:15:34 -0600
Quote:


On Feb 2, 2008, at 2:11 PM, John Von Essen <john at quonix.net> wrote:

Quote:
I posted an email a few days regarding a problem with hearing the
voicemail greeting on my sip phones.

It turns out to be a phone/stun/linksys issue - not an asterisk
issue.
Which brings up a couple of questions....

I always assumed that you can have multiple SIP phones behind a
Linksys
firewall/router (WRT54G) all using the same STUN server/port.

But apparently thats not the case. Is it a Linksys bug, a
Grandstream bug
in the BudgeTone-100 phone, or am I off base and just doing
something
wrong?

I cleary have problems as soon as I try to use a second phone behind
the
Linksys - registration issues, cant hear voicemail greeting, etc.,.

My next test was to run multiple STUN servers on the same machine
with
different ports. Then, for my multiple SIP phones behind the
Linksys, have
each phone use a different stun port.

Any thoughts?

John

I have 3 phones connected to 2 servers behind a 54g running openwrt
with no stun or any special configuration. I am running cisco phones
which do nat well natively.

-greg
Quote:

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com
--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
robert.norton at sopht...
Guest





PostPosted: Sat Feb 02, 2008 8:25 pm    Post subject: [asterisk-users] Multiple SIP phones behind a Linksys firewa Reply with quote

And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall?

In terms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio.

-----Original Message-----
From: Greg Oliver <greg.oliver at cistera.com>
Sent: Saturday, February 02, 2008 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall



On Feb 2, 2008, at 2:11 PM, John Von Essen <john at quonix.net> wrote:

Quote:
I posted an email a few days regarding a problem with hearing the
voicemail greeting on my sip phones.

It turns out to be a phone/stun/linksys issue - not an asterisk issue.
Which brings up a couple of questions....

I always assumed that you can have multiple SIP phones behind a
Linksys
firewall/router (WRT54G) all using the same STUN server/port.

But apparently thats not the case. Is it a Linksys bug, a
Grandstream bug
in the BudgeTone-100 phone, or am I off base and just doing something
wrong?

I cleary have problems as soon as I try to use a second phone behind
the
Linksys - registration issues, cant hear voicemail greeting, etc.,.

My next test was to run multiple STUN servers on the same machine with
different ports. Then, for my multiple SIP phones behind the
Linksys, have
each phone use a different stun port.

Any thoughts?

John

I have 3 phones connected to 2 servers behind a 54g running openwrt
with no stun or any special configuration. I am running cisco phones
which do nat well natively.

-greg
Quote:

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
lugosoft at gmail.com
Guest





PostPosted: Sat Feb 02, 2008 10:54 pm    Post subject: [asterisk-users] Multiple SIP phones behind a Linksys firewa Reply with quote

Quote:
I always assumed that you can have multiple SIP phones behind a Linksys
firewall/router (WRT54G) all using the same STUN server/port.

I got 10-20 SPA942's behind a OpenWRT router (on WRT54G, WRTSL54GS,
...) at several sites, no STUN, no special configuration, no problems
at all. Just as a precaution, I set the SIP port and RTP port range
for each phone differently so that it's unique (i.e. Phone 1 SIP port
6001 and RTP 10100-10199, etc.) but that's really just a precaution to
help the the Linux' conntrack on the OpenWRT a bit. It's not really
needed as the router will resolve port conflicts by rewriting the
ports transparently.

Bottom line, a few phones behind a well-behaved NAT should work just fine.

/Luki
Back to top
john at quonix.net
Guest





PostPosted: Sat Feb 02, 2008 11:22 pm    Post subject: [asterisk-users] Multiple SIP phones behind a Linksys firewa Reply with quote

The server is at a remote datacenter - no nat, no firewall, pure public
IP.

The phones are at home offices (i.e. DSL or Cable with Linksys-type
firewall/routers).

My initial testing was with a single SIP phone at the home office - and
everything worked fine. But when I have two SIP phones at the home office,
things start behaving badly.

I understand the issue of phone-to-phone, where both phones are behind a
nat at the home office - but that is not the issue I am having.

My main problem is when I have two phones at the home office, the second
phone cant register, and/or, you cant here the voicemail greeting when you
try to check messages.



----------------------------------------------------
Back to top
lists at minotaur.cc
Guest





PostPosted: Sun Feb 03, 2008 6:23 am    Post subject: [asterisk-users] Multiple SIP phones behind a Linksys firewa Reply with quote

Quote:
My main problem is when I have two phones at the home office, the second
phone cant register, and/or, you cant here the voicemail greeting when you
try to check messages.

I have seen this before on badly behaved home routers that have a hidden SIP Proxy, notably Zyxel wireless units. I've not seen it happening on either Linksys or Netgear units though.

Do you actually need STUN? In my experience it can cause more problems than it solves, especially if the public IP changes and the STUN server isn't due to be queried for another X seconds. If possible, and assuming it won't create unreasonable load on your * server, try dropping the registration interval down to something small like 300 (5 minutes), and disable STUN entirely (obviously making sure nat=yes is defined in sip.conf for those devices).

Regards,

Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons
Back to top
shadowym at hotmail.com
Guest





PostPosted: Sun Feb 03, 2008 2:10 pm    Post subject: [asterisk-users] Multiple SIP phones behind a Linksys firewa Reply with quote

Do you have a range of registration ports configured and forwarded through
the firewall on the server end? Ie. 5060-5065 for example.

On the Phone side you should forward 5060 to phone1 and 5061 to phone 2 etc.
and configure the phones to use that port for registration. You may need to
forward ports for the actual voice as well. 2 ports per phone so 10000-10001
for phone1 and 10002-10003 for phone2. It's either that or mess around with
STUN or Proxy servers or whatever.

SIP+NAT=headache



-----Original Message-----
From: john at quonix.net [mailto:john at quonix.net]
Sent: Saturday, February 02, 2008 8:23 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall

The server is at a remote datacenter - no nat, no firewall, pure public
IP.

The phones are at home offices (i.e. DSL or Cable with Linksys-type
firewall/routers).

My initial testing was with a single SIP phone at the home office - and
everything worked fine. But when I have two SIP phones at the home office,
things start behaving badly.

I understand the issue of phone-to-phone, where both phones are behind a
nat at the home office - but that is not the issue I am having.

My main problem is when I have two phones at the home office, the second
phone cant register, and/or, you cant here the voicemail greeting when you
try to check messages.
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services