Re: [asterisk-users] Asterisk on multi-homed systems

[Shlomo Dubrowin]

If I was wanted to multi-home on the same subnet I would use Ethernet
Bonding (similar to Windows Teaming) in a failover configuration.  This will
make one of the links on the LAN active and the second one as a failover in
case the first one goes down.  It takes a couple seconds for the 2nd link to
come up.  I am not using this in Asterisk at the moment, but I am using it
on other servers and it works great.  I don't know if this would drop a call
during failover, but it's something to explore.

  Shlomo


On 12/1/07, Steven [EMAIL PROTECTED] wrote:

 I have zero issues with multihomed asterisks.

 One potential issue is that some people are multihoming onto the same
 subnet.
 This will cause issues with many applications as normal routing usually
 sends data OUT the lower IP address if there are two on the
 same subnet.

 Multihoming, as a rule should be on separate network.

 My company's implementation is one three networks.
 One inside, One to ISP A and one to ISP B.

 Like I said, I have had zero issues.



 --
 --
 Steven

 http://www.connectech.org/



 Chris Bagnall [EMAIL PROTECTED] wrote in message
 news:[EMAIL PROTECTED]
  Greetings list,
 
  I remember a discussion many months ago which ISTR concluded that
 asterisk didn't play nicely at all in multi-homed setups (e.g.
  SIP packets not being sent out through the same interface they were
 received on, etc.).
 
  Is this still the case, or are there versions which have resolved the
 issue? Even if it's still the case, is this only a problem
  for SIP, or does it affect asterisk in general?
 
  I have a number of servers with dual NICs, each with an independent net
 connection. After a few recent failures with one provider,
  it'd be very useful to be able to use the other connection
 simultaneously, but only if it's not going to cause problems with the
  rest of the setup.
 
  Any suggestions gratefully appreciated.
 
  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
 
 
 
 
 
  ___
  --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

Re: [asterisk-users] Asterisk behind a PIX firewall?

[Shlomo Dubrowin]

Matt,

If your phone is using SIP, then you should enable sip inspection (7.x code
or above) or fixup sip (6.x code) and have a rule that allows source
(wherever you need) inbound on the outside interface to TCP 5060 (SIP
port).  The sip inspection or fixup should enable the proper ports for the
require RTP streams.  I had this working through an ASA at some point, but I
don't remember if both ends were doing NAT or only one end.  I don't know
the phone you are talking about, but you also might want to look into STUN
or ICE to get beyond the NAT Traversal issue, if that is what's causing the
problem.

In the Firewall log, are you seeing Denys? or drops?  Have you tried debug
sip on the firewall console?  I've been dealing with several ASA SIP issues
lately.  SIP trunking with NAT will certainly not work and there is a Cisco
Bug that my company discovered when setting up our PBX.

  Shlomo in Israel


On 11/27/07, Matt [EMAIL PROTECTED] wrote:

 Is there anything special that anyone here has had to do to get an Aastra
 phone (on the Internet) to talk to Asterisk behind a PIX firewall?

 Ports 1-2 UDP are open on the PIX and forwarding to the 
 Asteriskserver.   The
 Asterisk server's RTP.CONF is set to use 1-2.The phone
 registers, and will place AND receive calls, however, no audio is passed.
 The phone is an Aastra 9133i.

 ___
 --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