Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

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

[asterisk-users] Weird NAT issue ...


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
alan at blog-city.com
Guest





PostPosted: Mon Mar 17, 2008 8:59 am    Post subject: [asterisk-users] Weird NAT issue ... Reply with quote

Afternoon one and all.

I am having some interesting fun with our Asterisk setup.

We have two CISCO handsets (7960) sitting on the same network (NAT).

Each phone can successfully originate calls.

Each phone can be called successfully from outside

Each phone can be directly called by other extensions OUTSIDE the network
HOWEVER -- when those 2 phones try to call each other; the connection is
made, but no voice is heard.

Any advice as to where i need to look?

thanks

--
Alan Williamson

Try the free registration-less reminder service
http://www.yourli.st/

myBlog = 'http://alan.blog-city.com/';
Back to top
steve at one47.co.uk
Guest





PostPosted: Mon Mar 17, 2008 9:19 am    Post subject: [asterisk-users] Weird NAT issue ... Reply with quote

If the two phones attempt to refer to each other using their external
(NAT) IP addresses rather that their internal addresses, then it will
all go horribly wrong. You do not provide enough information about
asterisk IP addresses or firewalls for a possible solution, but
assuming you are using SIP and asterisk, you could try
"canreinvite=no" against the 2 phones to see if keeping the Asterisk
server in-the-loop helps.

Also look on the VoIP wiki for "externip" and "localnet" in the
sip.conf configuration.

Regards,
Steve

On Mon, Mar 17, 2008 at 1:59 PM, Alan Williamson <alan at blog-city.com> wrote:
Quote:
Afternoon one and all.

I am having some interesting fun with our Asterisk setup.

We have two CISCO handsets (7960) sitting on the same network (NAT).

Each phone can successfully originate calls.

Each phone can be called successfully from outside

Each phone can be directly called by other extensions OUTSIDE the network


HOWEVER -- when those 2 phones try to call each other; the connection is
made, but no voice is heard.

Any advice as to where i need to look?

thanks

--
Alan Williamson

Try the free registration-less reminder service
http://www.yourli.st/

myBlog = 'http://alan.blog-city.com/';

_______________________________________________
-- 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
anselm at hoffmeister-...
Guest





PostPosted: Mon Mar 17, 2008 9:40 am    Post subject: [asterisk-users] Weird NAT issue ... Reply with quote

Am Montag, den 17.03.2008, 13:59 +0000 schrieb Alan Williamson:
Quote:
Afternoon one and all.

I am having some interesting fun with our Asterisk setup.

We have two CISCO handsets (7960) sitting on the same network (NAT).

Each phone can successfully originate calls.

Each phone can be called successfully from outside

Each phone can be directly called by other extensions OUTSIDE the network


HOWEVER -- when those 2 phones try to call each other; the connection is
made, but no voice is heard.

Any advice as to where i need to look?

Hi Alan,

my guess is this has to do with the Audio path. As long as audio only
traverses the NAT router on the Cisco site, that device seems to handle
data paths quite well (you probably enabled different SIP ports for
those two devices? At least that helped me to a stable reachable phone,
which would just not work with more than one SIP 5060 phone behind a
single NAT).

The tricky part seems to be the "turnaround". One of the ciscos tries to
send audio data to the external ip address of the nat router, for the
other phone, and this might be something that the router does not
handle.

You could try to disallow direct audio between those two cisco phones by
forcing Astrisk to "stay in the audio path" (e.g. let all audio packets
go to asterisk, turnaround there and go to the other phone). This is
surely not optimal in bandwidth terms etc., but may solve such NAT
issues.

You can force Asterisk to stay in the audio path by specifying a Dial
option that requires Asterisk participation: Then it will not allow
direct connection automatically. Options requiring key presses (allow *
transfer or something, see Asterisk docs) should do.

Somehow the "reinvite" could have to do with that as well, but don't ask
me there Smile

BR,
Anselm
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