Hi all, I have my Asterisk behind a NAT router, but now it is configured to put that specific computer in DMZ (directly exposed to Internet). I intend to disable this and to open just the used ports. There is a list of TCP/UDP ports usd by Asterisk in order to connect to the outside world? One more question: When a call is established between an internal SIP phone (in LAN) and a phone from another place outside my router/firewall, using both the same codec (no conversion)... the call is still routed through the PBX or the PBX is used only for signaling and then a direct connection between the two phones is established? I ask this because if the audio stream is passed through the PBX then there is no need to open other ports on the firewall for the internal phones. Thanks, Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030524/10dcecf0/attachment.htm
from experience... the SIP is using asterisk as a proxy, so therefore is wont/cant handoff. On Sat, 24 May 2003 18:54:34 +0300, Dan wrote:>Hi all, > >I have my Asterisk behind a NAT router, but now it is configured to put that specific computer in DMZ (directly exposed to Internet). >I intend to disable this and to open just the used ports. >There is a list of TCP/UDP ports usd by Asterisk in order to connect to the outside world? > >One more question: When a call is established between an internal SIP phone (in LAN) and a phone from another place outside my router/firewall, using both the same codec (no conversion)... the call is still routed through the PBX or the PBX is used only for signaling and then a direct connection between the two phones is established? > >I ask this because if the audio stream is passed through the PBX then there is no need to open other ports on the firewall for the internal phones. > >Thanks, >Dan.
Hi Gary,> the SIP is using asterisk as a proxy, so therefore is wont/cant > handoff.Then how can it make codec conversion? I have a Cisco 7960 hardware SIP phone (with G.711) and an X-Lite (with GSM)... and they can talk each other through Asterisk.. Dan ----- Original Message ----- From: "Gary" <gary@ausmail.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, May 24, 2003 7:05 PM Subject: Re: [Asterisk-Users] TCP/UDP Ports used by Asterisk> from experience... > > the SIP is using asterisk as a proxy, so therefore is wont/cant > handoff. > > > > On Sat, 24 May 2003 18:54:34 +0300, Dan wrote: > > >Hi all, > > > >I have my Asterisk behind a NAT router, but now it is configured to putthat specific computer in DMZ (directly exposed to Internet).> >I intend to disable this and to open just the used ports. > >There is a list of TCP/UDP ports usd by Asterisk in order to connect tothe outside world?> > > >One more question: When a call is established between an internal SIPphone (in LAN) and a phone from another place outside my router/firewall, using both the same codec (no conversion)... the call is still routed through the PBX or the PBX is used only for signaling and then a direct connection between the two phones is established?> > > >I ask this because if the audio stream is passed through the PBX thenthere is no need to open other ports on the firewall for the internal phones.> > > >Thanks, > >Dan > > . > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > >
I said wont/cant handoff which means it always goes thru asterisk, there asterisk can to any codec conversion. Hey, I might be wriong, this is only what I have seen, and I stand to be corrected. (I would luv SIP to handoff (lisk IAX)), would save me a heap of bandwidth !! On Sat, 24 May 2003 19:10:38 +0300, Dan wrote:>Hi Gary, > >> the SIP is using asterisk as a proxy, so therefore is wont/cant >> handoff. >Then how can it make codec conversion? >I have a Cisco 7960 hardware SIP phone (with G.711) and an X-Lite (with >GSM)... and they can talk each other through Asterisk.. > >Dan > >----- Original Message ----- >From: "Gary" <gary@ausmail.com> >To: <asterisk-users@lists.digium.com> >Sent: Saturday, May 24, 2003 7:05 PM >Subject: Re: [Asterisk-Users] TCP/UDP Ports used by Asterisk > > >> from experience... >> >> the SIP is using asterisk as a proxy, so therefore is wont/cant >> handoff. >> >> >> >> On Sat, 24 May 2003 18:54:34 +0300, Dan wrote: >> >> >Hi all, >> > >> >I have my Asterisk behind a NAT router, but now it is configured to put >that specific computer in DMZ (directly exposed to Internet). >> >I intend to disable this and to open just the used ports. >> >There is a list of TCP/UDP ports usd by Asterisk in order to connect to >the outside world? >> > >> >One more question: When a call is established between an internal SIP >phone (in LAN) and a phone from another place outside my router/firewall, >using both the same codec (no conversion)... the call is still routed >through the PBX or the PBX is used only for signaling and then a direct >connection between the two phones is established? >> > >> >I ask this because if the audio stream is passed through the PBX then >there is no need to open other ports on the firewall for the internal >phones. >> > >> >Thanks, >> >Dan >> >> . >> >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users.
Robert Hajime Lanning
2003-May-24 09:25 UTC
[Asterisk-Users] TCP/UDP Ports used by Asterisk
<quote who="Gary">> I said wont/cant handoff > > which means it always goes thru asterisk, there asterisk can to any > codec conversion. > > Hey, I might be wriong, this is only what I have seen, and I stand to > be corrected. > > (I would luv SIP to handoff (lisk IAX)), would save me a heap of > bandwidth !!If the handoff is done, I would prefer it to be optional. I would hate to have to have all of my SIP/H.323 extensions out on the internet, instead of just the * server. Or to manage all of the inbound ports on the remote firewalls.
Sorry to understand something else.. Dan ----- Original Message ----- From: "Gary" <gary@ausmail.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, May 24, 2003 7:16 PM Subject: Re: [Asterisk-Users] TCP/UDP Ports used by Asterisk> I said wont/cant handoff > > which means it always goes thru asterisk, there asterisk can to any > codec conversion. > > Hey, I might be wriong, this is only what I have seen, and I stand to > be corrected. > > (I would luv SIP to handoff (lisk IAX)), would save me a heap of > bandwidth !! > > On Sat, 24 May 2003 19:10:38 +0300, Dan wrote: > > >Hi Gary, > > > >> the SIP is using asterisk as a proxy, so therefore is wont/cant > >> handoff. > >Then how can it make codec conversion? > >I have a Cisco 7960 hardware SIP phone (with G.711) and an X-Lite (with > >GSM)... and they can talk each other through Asterisk.. > > > >Dan > > > >----- Original Message ----- > >From: "Gary" <gary@ausmail.com> > >To: <asterisk-users@lists.digium.com> > >Sent: Saturday, May 24, 2003 7:05 PM > >Subject: Re: [Asterisk-Users] TCP/UDP Ports used by Asterisk > > > > > >> from experience... > >> > >> the SIP is using asterisk as a proxy, so therefore is wont/cant > >> handoff. > >> > >> > >> > >> On Sat, 24 May 2003 18:54:34 +0300, Dan wrote: > >> > >> >Hi all, > >> > > >> >I have my Asterisk behind a NAT router, but now it is configured toput> >that specific computer in DMZ (directly exposed to Internet). > >> >I intend to disable this and to open just the used ports. > >> >There is a list of TCP/UDP ports usd by Asterisk in order to connectto> >the outside world? > >> > > >> >One more question: When a call is established between an internal SIP > >phone (in LAN) and a phone from another place outside my router/firewall, > >using both the same codec (no conversion)... the call is still routed > >through the PBX or the PBX is used only for signaling and then a direct > >connection between the two phones is established? > >> > > >> >I ask this because if the audio stream is passed through the PBX then > >there is no need to open other ports on the firewall for the internal > >phones. > >> > > >> >Thanks, > >> >Dan > >> > >> . > >> > >> > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > >> Asterisk-Users@lists.digium.com > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > > > > > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > > . > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
I believe it does handoff. I do it all the time and for me is extremely helpful since I don?t want all the calls to go through my * box. To do a handoff you just need to set canreinvite=yes on the sip.conf . The problem I found then is with the ATA doing NAT since you need to set canreinvite=no, and therefore it won?t hand it off. ----- Original Message ----- From: "Gary" <gary@ausmail.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, May 24, 2003 1:16 PM Subject: Re: [Asterisk-Users] TCP/UDP Ports used by Asterisk> I said wont/cant handoff > > which means it always goes thru asterisk, there asterisk can to any > codec conversion. > > Hey, I might be wriong, this is only what I have seen, and I stand to > be corrected. > > (I would luv SIP to handoff (lisk IAX)), would save me a heap of > bandwidth !! > > On Sat, 24 May 2003 19:10:38 +0300, Dan wrote: > > >Hi Gary, > > > >> the SIP is using asterisk as a proxy, so therefore is wont/cant > >> handoff. > >Then how can it make codec conversion? > >I have a Cisco 7960 hardware SIP phone (with G.711) and an X-Lite (with > >GSM)... and they can talk each other through Asterisk.. > > > >Dan > > > >----- Original Message ----- > >From: "Gary" <gary@ausmail.com> > >To: <asterisk-users@lists.digium.com> > >Sent: Saturday, May 24, 2003 7:05 PM > >Subject: Re: [Asterisk-Users] TCP/UDP Ports used by Asterisk > > > > > >> from experience... > >> > >> the SIP is using asterisk as a proxy, so therefore is wont/cant > >> handoff. > >> > >> > >> > >> On Sat, 24 May 2003 18:54:34 +0300, Dan wrote: > >> > >> >Hi all, > >> > > >> >I have my Asterisk behind a NAT router, but now it is configured toput> >that specific computer in DMZ (directly exposed to Internet). > >> >I intend to disable this and to open just the used ports. > >> >There is a list of TCP/UDP ports usd by Asterisk in order to connectto> >the outside world? > >> > > >> >One more question: When a call is established between an internal SIP > >phone (in LAN) and a phone from another place outside my router/firewall, > >using both the same codec (no conversion)... the call is still routed > >through the PBX or the PBX is used only for signaling and then a direct > >connection between the two phones is established? > >> > > >> >I ask this because if the audio stream is passed through the PBX then > >there is no need to open other ports on the firewall for the internal > >phones. > >> > > >> >Thanks, > >> >Dan > >> > >> . > >> > >> > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > >> Asterisk-Users@lists.digium.com > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > > > > > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > > . > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >