I have a need for a large number of remote phones. I want to use GSM between the phones and the conversion server which will transcode to ulaw eventually send or recieve calls via the PSTN (ulaw). I am curious is anyone has any ideas on the easiest way to create a dedicated codec conversion box. It will be running openvpn and so will the remote PCs with softphones (x-lite). So I want the remote softphones to connect to the codec conversion asterisk box and then send the call to the main Asterisk server as ulaw and pass call in and out the pstn as ulaw. Any ideas for a simple implementation without creating all kinds of funky conf files. Seems simple but the solution eludes me (maybe because I have been working over 18 hours. Thanks, Steve Totaro
Klaverstyn, David C
2007-Oct-25 04:20 UTC
[asterisk-users] Dedicated Codec Conversion Server
The way I would accomplish this is to have 2 Asterisk boxes. Your
conversion server would just have a dial plan to forward all calls to
the Asterisk box that has the PSTN interface. Once the PSTN Asterisk
Server receives the calls it just routes the call based on dial plan
rules.
{Internet -> VPN} -> Conversion Server -> Asterisk PSTN.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
Totaro
Sent: Thursday, 25 October 2007 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dedicated Codec Conversion Server
I have a need for a large number of remote phones. I want to use GSM
between the phones and the conversion server which will transcode to
ulaw eventually send or recieve calls via the PSTN (ulaw).
I am curious is anyone has any ideas on the easiest way to create a
dedicated codec conversion box. It will be running openvpn and so will
the remote PCs with softphones (x-lite).
So I want the remote softphones to connect to the codec conversion
asterisk box and then send the call to the main Asterisk server as ulaw
and pass call in and out the pstn as ulaw.
Any ideas for a simple implementation without creating all kinds of
funky conf files. Seems simple but the solution eludes me (maybe
because I have been working over 18 hours.
Thanks,
Steve Totaro
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Um, yeah, the part you suggest is obvious. What about having two different SIP accounts for each phone, I guess I need to do one for inbound and one for outbound? Different extensions or whatever. Thanks, Steve Klaverstyn, David C wrote:> The way I would accomplish this is to have 2 Asterisk boxes. Your > conversion server would just have a dial plan to forward all calls to > the Asterisk box that has the PSTN interface. Once the PSTN Asterisk > Server receives the calls it just routes the call based on dial plan > rules. > > {Internet -> VPN} -> Conversion Server -> Asterisk PSTN. > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve > Totaro > Sent: Thursday, 25 October 2007 1:54 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Dedicated Codec Conversion Server > > I have a need for a large number of remote phones. I want to use GSM > between the phones and the conversion server which will transcode to > ulaw eventually send or recieve calls via the PSTN (ulaw). > > I am curious is anyone has any ideas on the easiest way to create a > dedicated codec conversion box. It will be running openvpn and so will > the remote PCs with softphones (x-lite). > > So I want the remote softphones to connect to the codec conversion > asterisk box and then send the call to the main Asterisk server as ulaw > and pass call in and out the pstn as ulaw. > > Any ideas for a simple implementation without creating all kinds of > funky conf files. Seems simple but the solution eludes me (maybe > because I have been working over 18 hours. > > Thanks, > Steve Totaro > >
Will "All" inbound calls go to SIP phones on the conversion server? If so, you just need a .X extension that forwards all inbound calls to the second server. But, thinking about this, it would appear that the second "conversion" server will need most of your dialplan, and as such may still be affected by codec translation load. Is that what you are trying to avoid? I am looking at a similar design, but my goal is not to offload the codec translation, but to offload the TDM functions. This seems like it will be quite easy. as the server with TDM function will have a very simple dialplan. But, my "extension" server will be just as complex as it is today. (but IP only) I am not sure you can offload codec translation without a very complex design. -- -- Steven http://www.glimasoutheast.org "Steve Totaro" <stotaro at first-notification.com> wrote in message news:47209177.3030805 at first-notification.com...> Um, yeah, the part you suggest is obvious. > > What about having two different SIP accounts for each phone, I guess I > need to do one for inbound and one for outbound? Different extensions > or whatever. > > Thanks, > Steve > > Klaverstyn, David C wrote: >> The way I would accomplish this is to have 2 Asterisk boxes. Your >> conversion server would just have a dial plan to forward all calls to >> the Asterisk box that has the PSTN interface. Once the PSTN Asterisk >> Server receives the calls it just routes the call based on dial plan >> rules. >> >> {Internet -> VPN} -> Conversion Server -> Asterisk PSTN. >> >> -----Original Message----- >> From: asterisk-users-bounces at lists.digium.com >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve >> Totaro >> Sent: Thursday, 25 October 2007 1:54 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: [asterisk-users] Dedicated Codec Conversion Server >> >> I have a need for a large number of remote phones. I want to use GSM >> between the phones and the conversion server which will transcode to >> ulaw eventually send or recieve calls via the PSTN (ulaw). >> >> I am curious is anyone has any ideas on the easiest way to create a >> dedicated codec conversion box. It will be running openvpn and so will >> the remote PCs with softphones (x-lite). >> >> So I want the remote softphones to connect to the codec conversion >> asterisk box and then send the call to the main Asterisk server as ulaw >> and pass call in and out the pstn as ulaw. >> >> Any ideas for a simple implementation without creating all kinds of >> funky conf files. Seems simple but the solution eludes me (maybe >> because I have been working over 18 hours. >> >> Thanks, >> Steve Totaro >> >> > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >