my config asterisk server ----2wirerouter ----internet ---- other router ----- grandstream budgettone 101 IP phone on my ip Phone i put sip server my public ip (is mapped to asterisk server on 5060 y 5004 port numbers) use nat transversal = yes and stun server = stun.xten.com<http://stun.xten.com> the extension register ok on asterisk server , but not audio is transmited on answer a call how i can solve this. thanks -- David Romero ################################## -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050728/5d5d6e15/attachment.htm
On 7/29/05, David Romero <romdav@gmail.com> wrote:> my config > > asterisk server ----2wirerouter ----internet ---- other router ----- > grandstream budgettone 101 IP phone > > on my ip Phone i put > sip server my public ip (is mapped to asterisk server on 5060 y 5004 > port numbers) > use nat transversal = yes and stun server = stun.xten.com > > the extension register ok on asterisk server , but not audio is transmited > on answer a call > > how i can solve this.Check on codec mismatch. Holden
now i forward ports 10000-20000 to my asterisk server but the problem is the same. i not understand wy not work ---------- Forwarded message ---------- From: Holden Hao <holdenhao@gmail.com> Date: Jul 28, 2005 9:09 PM Subject: Re: [Asterisk-Users] Nat Transversal To: David Romero <romdav@gmail.com> On 7/29/05, David Romero <romdav@gmail.com> wrote:> i try whit other codec but not work. > > i try the phone on other site, > and work nice just one time, i not change anyting and reboot the phone > after reboot not work anymore, if change the public ip address of my > router the phone work again just one time > how i can fix it?.Your Asterisk has a private IP, right? Check that you have properly forwarded all the ports required. Apart from 5060 for SIP your need to port forward the RTP ports 10000-20000 to your asterisk server. Holden -- David Romero ################################## -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050729/10079d47/attachment.htm
the stun server does not work with Symmetric NAT traversal..... do you know what kind of NAT do you have? this stun server try to detect youre NAT type http://sourceforge.net/projects/stun/ more info about the problems with stun and nat http://www.newport-networks.com/whitepapers/fwnatwpes3.html not sure if its suitable to your problem, hope it helps. What does asterisk console says when the call starts? best regards On 7/28/05, David Romero <romdav@gmail.com> wrote:> my config > > asterisk server ----2wirerouter ----internet ---- other router ----- > grandstream budgettone 101 IP phone > > on my ip Phone i put > sip server my public ip (is mapped to asterisk server on 5060 y 5004 > port numbers) > use nat transversal = yes and stun server = stun.xten.com > > the extension register ok on asterisk server , but not audio is transmited > on answer a call > > how i can solve this. > > thanks > -- > David Romero > ################################## > _______________________________________________ > 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 > >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"
Have a look at this tutorial about SIP and NAT problems, it might help you... http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html Zoa David Romero wrote:> now i forward ports 10000-20000 to my asterisk server but the problem > is the same. > i not understand wy not work > > > ---------- Forwarded message ---------- > From: *Holden Hao* <holdenhao@gmail.com <mailto:holdenhao@gmail.com>> > Date: Jul 28, 2005 9:09 PM > Subject: Re: [Asterisk-Users] Nat Transversal > To: David Romero <romdav@gmail.com <mailto:romdav@gmail.com>> > > On 7/29/05, David Romero <romdav@gmail.com <mailto:romdav@gmail.com>> > wrote: > > i try whit other codec but not work. > > > > i try the phone on other site, > > and work nice just one time, i not change anyting and reboot the phone > > after reboot not work anymore, if change the public ip address of my > > router the phone work again just one time > > how i can fix it?. > > Your Asterisk has a private IP, right? Check that you have properly > forwarded all the ports required. Apart from 5060 for SIP your need > to port forward the RTP ports 10000-20000 to your asterisk server. > > > Holden > > > -- > David Romero > ################################## > >------------------------------------------------------------------------ > >_______________________________________________ >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 -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 254 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/89a0c115/signature.pgp