[Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

[Bharath]

Manny,
Sorry if my post caused any confusion. I'm talking about 2 different
locations of the server & client.
My Asterisk server is located at my office and is not behind a NAT or
firewall. It is directly connected to my Cable modem.
I'm using a Sipura2002 ATA at home. This ATA is connected to the asterisk
server which is located at my office. The ATA at my home is behind a NAT.
The ATA sucessfully registers and can also make & recieve calls only the
voice is blocked.
The  external ports 10000-20000 were not opened on my Asterisk box. Only
port 5060-5082 were opened. I guess thats the reason I was not able to hear
any voice. Will try that this evening and post my results.

Thanks


On 11/23/05, Manny A. Wise <mannywise at gmail.com> wrote:
>
>  Well, as the user stated on the original message, the asterisk server is
> behind a NAT and the client is also behind a NAT….
>
> if you make it work just by opening ports, let me know..I have never been
> able to get it to work, that's why I don't use sip, just plain iax2 for
> everything… J
>
>
>
> Manny
>
>
>
> -----Original Message-----
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Bharath
> *Sent:* Wednesday, November 23, 2005 10:08 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a
> public domain
>
> Thanks Michael,
> I think thats is the problem, I have opened only ports 5060-5082, I need
> to open 10000-20000 as well. I will try that and post the result when i get
> back home.
> Thanks
>
> On 11/23/05, *Michael West* <mwest at westmarkinc.com> wrote:
>
> I'm pasting something from another user on this list from 14/11/05
>
> I would recommend that you do a little research on google, voip- info.org,
> and the list archives.
>
> To connect to an Asterisk box that sits behind NAT, you need to forward
> ports 5060 and 10000-20000 too the asterisk box, and you need to configure
> the externip, localnet, and nat variables in sip.conf.
>
> audio problems are almost always due to the RTP stream (ports 10000-20000)
> not being forwarded properly, either due to the port forwarding setup or the
> sip.conf settings.
>
> Tom
>
> ----------------------------------------------------------
>
> Tom Rymes
>
> Cascade Link Systems
>
> *www.cascadelinksystems.com*
>
> (603) 375-1414
>
>
>  ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Bharath
> Khambadkone
> *Sent:* Wednesday, November 23, 2005 9:29 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a
> public domain
>
> By default AMP had NAT=yes in sip.conf, I read in some posts to change it
> to one, i was just trying my luck if that works. I have tried NAT=yes, The
> Phone gets registered, I can also make & recieve calls but as soon as the
> call is picked I dont hear anything at both ends. Does this have anything to
> do with codecs?
>
> Thanks
>
> On 11/22/05, *C F* <shmaltz at gmail.com> wrote:
>
> On 11/22/05, Bharath Khambadkone <bkalthod at gmail.com> wrote:
> > Hello All,
> >  I'm fairly new to asterisk. I have read about the problems about NAT,
> But
> > can't seem to find a solution.
> >  My Asterisk is on a public domain, there is no NAT or firewall in front
> of
>
>
> If no nat then why do you have nat=1 in sip.conf?
>
>
> > the asteris box. I have sucessfully connected iax2 softphones & was able
> to
> > recieve & make calls. In the same locations where I have the iax2
> extensions
> > working I have set up a a SIP softphone & a SIP ATA (Sipura2002). Both
> teh
> > sip phones are able to register. I can also make & recieve calls but
> cannot
> > hear anything after the call is answered at both ends. I'm not sure what
> is
> > causing this problem. By the way I'm using SME server 7(centos 4.2
> )  with
> > A at H installed.
> >
> >  my Sip.conf :
> >  [2008] ;(Sipura2002)
> >  username=2008
> >  type=friend
> >  secret=2008
> >  record_out=Adhoc
> >  record_in=Adhoc
> >  qualify=no
> >  port=5060
> >  nat=1
> >  mailbox=2008 at device
> >  host=dynamic
> >  dtmfmode=rfc2833
> >  context=from-internal
> >  canreinvite=no
> >  callerid=device <2008>
> >
> >
> >  [2009] ;X-Lite Soft Phone
> >  username=2009
> >  type=friend
> >  secret=2009
> >  record_out=Adhoc
> >  record_in=Adhoc
> >  qualify=no
> >  port=5060
> >  nat=1
> >  mailbox=2009 at device
> >  host=dynamic
> >  dtmfmode=rfc2833
> >  context=from-internal
> >  canreinvite=no
> >  callerid=device <2009>
> >
> >  Thanks in advance..
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051123/af18cf1b/attachment.htm