[Asterisk-Users] SIP Proxy Problem (NAT Environment)

[Michael Shuler]

Then your firewall is closing the return RTP port to fast.  Check for the
latest firmware and also make sure the SPI (stateful packet inspection) is
turned off if you router has that option.  Otherwise you may have to give up
and fall back to port forwarding.

----------------------------------------

Michael Shuler, C.E.O.
BitWise Systems, Inc.
1301 W. Pioneer Parkway
Peoria, IL 61615
Office: (217) 585-0357
Cell: (309) 657-6365
Fax: (309) 213-3500
E-Mail: mike at bwsys.net
Customer Service: (877) 976-0711 

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Markus
Miertschink
Sent: Tuesday, April 06, 2004 10:58 AM
To: asterisk-users at lists.digium.com
Subject: AW: [Asterisk-Users] SIP Proxy Problem (NAT Environment)



It is set to yes.

Strangely it works - only if I make the call from one direction one voice
channel gets no voice transmitted.

 


  _____  


Von: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] Im Auftrag von Michael Shuler
Gesendet: Dienstag, 6. April 2004 17:30
An: asterisk-users at lists.digium.com
Betreff: RE: [Asterisk-Users] SIP Proxy Problem (NAT Environment)

 

Make sure you have nat=yes for the sip.conf entry for SIPGATE.

 

 

----------------------------------------

Michael Shuler, C.E.O.
BitWise Systems, Inc.
1301 W. Pioneer Parkway
Peoria, IL 61615
Office: (217) 585-0357
Cell: (309) 657-6365
Fax: (309) 213-3500
E-Mail: mike at bwsys.net
Customer Service: (877) 976-0711 

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Markus
Miertschink
Sent: Tuesday, April 06, 2004 10:09 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] SIP Proxy Problem (NAT Environment)

I have the problem calling from a NAT firewalled SIP Phone to sipgate.de.
Somehow it is working. The connection works. Everything seems to be fine. I
could talk to the external phone. But: the external voice gets not
transmitted to my internal (calling) phone.

 

Calling inbound from SipGate to the same phone works perfectly. Both
channels are perfectly "proxied".

 

The * is used as a proxy and registered the sipgate account.

 

This way works:

 

SIPGATE ----->NAT(IP0)------>Asterisk(IP1)------->IP-Phone(IP2)

 

This not (partially):

 

SIPGATE<----NAT(IP0)<-----Asterisk(IP1)<-----IP-Phone(IP2)

 

All ports are mapped to the address of IP1. If I look into the tcpdump log
it seems that all ports used there are matching my NAT settings. Asterisk is
happy. No problems.

I only run into immediate abortion of the call if Asterisk is configured
using outbound_address with the ip of IP0.

 

I have no clue what to do anymore.

 

Regards,

Markus

 

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