Bharath Khambadkone
2005-Nov-21 22:28 UTC
[Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
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 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@Hinstalled. 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@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@device host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device <2009> Thanks in advance.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051121/da753646/attachment.htm
Alexander Lopez
2005-Nov-21 22:35 UTC
[Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
change nat=1 to nat=yes ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Bharath Khambadkone Sent: Tue 11/22/2005 12:28 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain 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 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@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@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@device host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device <2009> Thanks in advance.. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4554 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051121/c54ecf73/attachment.bin
C F
2005-Nov-22 09:58 UTC
[Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
On 11/22/05, Bharath Khambadkone <bkalthod@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 ofIf 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@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@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@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@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 > >
Michael West
2005-Nov-23 07:52 UTC
[Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
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 <outbind://12/www.cascadelinksystems.com> (603) 375-1414 ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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@gmail.com> wrote: On 11/22/05, Bharath Khambadkone <bkalthod@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@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@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@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@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 > > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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/a35c78d7/attachment.htm
Tom Rymes
2005-Nov-25 20:54 UTC
[Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
On Nov 25, 2005, at 7:00 PM, Manny A. Wise wrote:> -----Original Message----- > From: Tom Rymes [mailto:trymes@cascadelinksystems.com][snip]>> On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote: >> [snip] >>> 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 >> >> Manny, >> >> I have this working as I write this. (I just hung up the phone.) In >> fact, I brought a Cisco 7940G to a completely unknown nat-ed network >> the other day, plugged it in and started making calls right away. >> Here's the setup I have for this specific configuration: >> >> 1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but >> it's still NAT. I just don't have to forward ports this way) >> 2.) externip, localnet, nat settings configured in the sip.conf file >> (sip_nat.conf for Asterisk@Home) >> 3.) Cisco phone (or whatever SIP UA you choose) configured for NAT >> (via the SIP<MAC>.cnf file for Cisco) >> 4.) Lather, rinse, repeat if necessary >> >> Hopefully that will work for you. I'd rather use IAX and avoid these >> problems altogether, but I have yet to find an IAX hardphone I am >> willing to use. In fact, for softphone use, I do indeed use IAX via >> LoudHush for the mac. (Great piece of software, BTW. No connection >> here, just a happy user...) >> >> Tom > > Great!!, this did the trick, now we have audio... > We are using a Sipura 2000 for testing.... > The Sipura now can call out and have audio...the only problem left > is that > the sipura can't receive calls, when the extension is dialed, the > recording > says, the person is on the phone.....any ideas??? > > I changed the externip=, localnet= and nat=yes in sip.com and in the > extension setup in amp nat=1...... missing anything???? > > THANKS!!!!!!!!!!!!!!!! > > MannyIt sounds as if your extension isn't registered. Make sure that the extension is configured as dynamic in sip.conf (or AMP) and as nat=yes. Also, make sure that the Sipura is configured through its web interface to register and it has the right user and password entered. Once this is done, when you type 'sip show peers' from the CLI your Sipura's extension should be listed, and show a 'D' and an 'N' for dynamic and nat. Also, it sounds like you are using AMP and or A@H, so make sure that you put the nat, externip, and localnet parameters in the sip_nat.conf file, *NOT* the sip.conf file, as that is likely to get overwritten by AMP. From my installation (obviously, substitute your external IP for the xxx.xxx.xxx.xxx below...): [root@mercury root]# cat /etc/asterisk/sip_nat.conf nat=yes externip=xxx.xxx.xxx.xxx localnet=10.0.0.0/255.255.255.0 Other than that, I recommend further google and voip-info spelunking expeditions to track down your problem. I think that voxilla.com also has good resources on the Sipuras Tom -------------------- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 "Intelligent technology solutions for small businesses."