Hello List, I set up Asterisk for a client. He is using Bellsouth DSL and is behind a Linksys router. I opend all the ports. (5000-600 and 10000-20000). For some reson no one from the out side can connect in. I want to know if anyone had a problem with either Linksys routers or Bell South business DSL. Thanks. David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050911/e25330f2/attachment.htm
5000-600? Do you mean 5060? That is the port for 5060. 10000-20000 is for RTP. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dovid B. Asterisk Users Sent: Sunday, September 11, 2005 12:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Connection Problems Hello List, I set up Asterisk for a client. He is using Bellsouth DSL and is behind a Linksys router. I opend all the ports. (5000-600 and 10000-20000). For some reson no one from the out side can connect in. I want to know if anyone had a problem with either Linksys routers or Bell South business DSL. Thanks. David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050911/397c693e/attachment.htm
Are you using the Linksys router as your PPPoE termination or are using the Netopia?? Alex -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dovid B. Asterisk Users Sent: Sunday, September 11, 2005 3:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Connection Problems Hello List, I set up Asterisk for a client. He is using Bellsouth DSL and is behind a Linksys router. I opend all the ports. (5000-600 and 10000-20000). For some reson no one from the out side can connect in. I want to know if anyone had a problem with either Linksys routers or Bell South business DSL. Thanks. David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050911/80404fee/attachment.htm
Hi All
I have a Cisco 7960 which is connected remotely to an Asterisk server.
Both are unfortunately behind NAT.
The Phone registers and is show in sip show peers, with the correct
public ip for the phone and a 100ms qualify time
(1) I can dial the phone from another phone, it will ring but no voice
goes through in fact I get this error on * console
SIP response 481 "Call Leg/Transaction Does Not Exist
(2) the phone can make calls outbound fine, with voice no problems ?
http://channels.debian.net/paste/3409 holds the SIP debug for the
phone extension 650
port forwarding is also set on both sides, and sip.conf has the nat=yes
, externip and localnet all set correctly
Thank always
Barry
I had issues using NAT when having multiple phones as well as single phones behind NAT. You can try setting port forwarding on the phones side as well as look at a better router. Some routers will make you pull your hair out while others will work almost perfectly (this explains my now bald head :) ) Dovid ----- Original Message ----- From: "Barry Fawthrop" <barry@ttienterprises.org> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Saturday, August 12, 2006 7:27 AM Subject: [asterisk-users] SIP Connection Problems> Hi All > > I have a Cisco 7960 which is connected remotely to an Asterisk server. > > Both are unfortunately behind NAT. > The Phone registers and is show in sip show peers, with the correct public > ip for the phone and a 100ms qualify time > > (1) I can dial the phone from another phone, it will ring but no voice > goes through in fact I get this error on * console > SIP response 481 "Call Leg/Transaction Does Not Exist > > (2) the phone can make calls outbound fine, with voice no problems ? > > http://channels.debian.net/paste/3409 holds the SIP debug for the phone > extension 650 > > port forwarding is also set on both sides, and sip.conf has the nat=yes , > externip and localnet all set correctly > > > Thank always > > Barry > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Thanks Dovid I have port forwarding enabled on the linksys router ports 5060 and 10000-20000. I was wondering if I should also enable DMZ to the internal IP address of the phone ? Thanks Barry Dovid Bender wrote:> I had issues using NAT when having multiple phones as well as single > phones behind NAT. You can try setting port forwarding on the phones > side as well as look at a better router. Some routers will make you > pull your hair out while others will work almost perfectly (this > explains my now bald head :) ) > > Dovid > > ----- Original Message ----- From: "Barry Fawthrop" > <barry@ttienterprises.org> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Saturday, August 12, 2006 7:27 AM > Subject: [asterisk-users] SIP Connection Problems > >
Yes. It should help. See what happens. Also it can be your router at either or both ends. ----- Original Message ----- From: "Barry Fawthrop" <barry@ttienterprises.org> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, August 13, 2006 9:23 AM Subject: Re: [asterisk-users] SIP Connection Problems> Thanks Dovid > > I have port forwarding enabled on the linksys router ports 5060 and > 10000-20000. I was wondering if I should also enable DMZ to the internal > IP address of the phone ? > > Thanks > Barry > > Dovid Bender wrote: >> I had issues using NAT when having multiple phones as well as single >> phones behind NAT. You can try setting port forwarding on the phones side >> as well as look at a better router. Some routers will make you pull your >> hair out while others will work almost perfectly (this explains my now >> bald head :) ) >> >> Dovid >> >> ----- Original Message ----- From: "Barry Fawthrop" >> <barry@ttienterprises.org> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >> <asterisk-users@lists.digium.com> >> Sent: Saturday, August 12, 2006 7:27 AM >> Subject: [asterisk-users] SIP Connection Problems >> >> > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >