Re: [asterisk-users] Asterisk NAT
Anyway, thank you so much. ;-) On Fri, Feb 21, 2014 at 9:32 PM, Rusty Newton wrote: > On Fri, Feb 21, 2014 at 11:13 AM, Gholamreza Sabery > wrote: > > > > Dear Mr. Newton > > Thank you for your response. I red the wiki and sip.conf sample file and > I > > know about directmedia option. Actually these options are for times that > you > > know about your connected networks (you know which clients are behind NAT > > and which are not). But my configuration is different. I have an > A2Billing > > From my understanding and the documentation, the intent with > directmedia=nonat is that it will act like directmedia=yes if the peer > is detected as *not* being behind NAT, and directmedia=no if the peer > is detected as being behind NAT. This implies that the administrator > would not know ahead of time what is needed, otherwise seemingly you > would just use yes or no. However I'm still not sure that will be > helpful for your particular scenario. > > > users. Here I want Asterisk to automatically detect when two users are > > behind the same NAT and redirect their traffic inside that NAT; this way > the > > load of RTP traffic on Asterisk server will be reduced. > > I don't know that this is possible with any simple Asterisk > configuration. Good luck! > > -- > Rusty Newton > Digium, Inc. | Community Support Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct: +1 256 428 6200 > > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk NAT
On Fri, Feb 21, 2014 at 11:13 AM, Gholamreza Sabery wrote: > > Dear Mr. Newton > Thank you for your response. I red the wiki and sip.conf sample file and I > know about directmedia option. Actually these options are for times that you > know about your connected networks (you know which clients are behind NAT > and which are not). But my configuration is different. I have an A2Billing >From my understanding and the documentation, the intent with directmedia=nonat is that it will act like directmedia=yes if the peer is detected as *not* being behind NAT, and directmedia=no if the peer is detected as being behind NAT. This implies that the administrator would not know ahead of time what is needed, otherwise seemingly you would just use yes or no. However I'm still not sure that will be helpful for your particular scenario. > users. Here I want Asterisk to automatically detect when two users are > behind the same NAT and redirect their traffic inside that NAT; this way the > load of RTP traffic on Asterisk server will be reduced. I don't know that this is possible with any simple Asterisk configuration. Good luck! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk NAT
Dear Mr. Newton Thank you for your response. I red the wiki and sip.conf sample file and I know about directmedia option. Actually these options are for times that you know about your connected networks (you know which clients are behind NAT and which are not). But my configuration is different. I have an A2Billing server + Asterisk (these two share a database using ARA); I also wrote a web service that allows users to automatically register and get a username and password. After registration users can connect to Asterisk to call other users. Here I want Asterisk to automatically detect when two users are behind the same NAT and redirect their traffic inside that NAT; this way the load of RTP traffic on Asterisk server will be reduced. On Thu, Feb 20, 2014 at 11:13 PM, Rusty Newton wrote: > On Wed, Feb 19, 2014 at 2:55 AM, A J Stiles > wrote: > > On Wednesday 19 Feb 2014, Gholamreza Sabery wrote: > >> Hello, a few days ago I sent a question: > >> > >> > http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html > >> > >> but no one answered me! I just want to know is it possible or not? > > > > No answer on the list probably just means the question was answered > before; so > > your best bet is to search the mailing list archives and the wiki at > > http://voip-info.org > > Eventually, you will have been yomping around in Tech Land for long > enough to > > graduate from "ignorant tourist" to "seasoned traveller" -- and then you > get > > to ignore noob questions yourself. Or set yourself up as a tour guide, > if you > > feel that way inclined :) > > It is worth nothing that the official Asterisk wiki is at > http://wiki.asterisk.org. If there is something missing from there, > feel free to let me or someone in #asterisk-dev know and we'll make > sure things get updated. One thing I do have on my to-do list is a NAT > guide. > > -- > Rusty Newton > Digium, Inc. | Community Support Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct: +1 256 428 6200 > > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cancel a ringing SIP call when the other party disconnect
Hi, Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged). But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with the qualify. Is there any setup or asterisk configuration I need to enable to have A close its call ? Note: when A is already talking with B, the call is hanged up on rtp timeout. But not during the Ringing phase. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a client: can I get the remote SIP server to ignore rport?
Am 21.02.2014 15:12, schrieb Andres: Wow, if this is the case then I would be changing VM providers immediately.You would have problems not only with Asterisk but with most other services you wanted to host on it. There are many VM providers out there that work just fine with Asterisk even on a 1:1 NAT like Amazon Web Services. Update: the VPS provider has fixed the issue with their router and now everything works as expected. Unfortunately, in this particular "exotic" country, it seems there is exactly *1* VPS hosting provider, so switching to another provider would not be really feasible, and switching to another country is not at option either. :) All is good now! Thanks! Markus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a client: can I get the remote SIP server to ignore rport?
And I'm pretty sure if you look at any of those peers that have a non-5060 port, the routers in front of them will rewrite packets destined for ports 53277, 4121, 47822 etc. to the proper corresponding internal IP:port where something is listening. The router of my provider won't. It rewrites ports on outgoing packets, but it passes incoming packets 1:1 to the VM. Wow, if this is the case then I would be changing VM providers immediately.You would have problems not only with Asterisk but with most other services you wanted to host on it. There are many VM providers out there that work just fine with Asterisk even on a 1:1 NAT like Amazon Web Services. IMHO, my hosting provider is at fault, and I'm working with them to get it fixed. I was just wondering if there is some magic switch which can fix such a broken scenario. Thank you! Markus -- Technical Support http://www.cellroute.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variables are empty after Redirecting a channel
Sorry there is a mistake, actually I don't redirect executing channel:
exten => s,n,ChannelRedirect(${CHANNEL_TO_REDIRECT},context_2,AMD,1)
I can't use Goto because the ${CHANNEL_TO_REDIRECT} is active channel and
Goto drops (hangs-up) it.
So is there a way to pass variable to the target context after
ChannelRedirect?
Thanks,
Igor
On Thu, Feb 20, 2014 at 2:37 PM, Joshua Colp wrote:
>
> You should be using Goto here instead of ChannelRedirect, since you are
> redirecting the channel which is executing ChannelRedirect (that
> slightly made my head hurt). Switching should also make the variable
> work as you desire.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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