It's early no need to be on the ball yet. In my past experiences with g729 and g711, codec negotiation is less than perfect. If you support both and want to prefer one and then you send to a device that only supports the other, I've had them reject the call even though it should have worked perfectly. I couldn't find any rhyme or reason for it either, it may be different if you're using a provider with your soft phone with no asterisk involved though.
So much R&D, so little time. - Ian On 7/13/06, Alex Robar <[EMAIL PROTECTED]> wrote:
Wow, I'm really not on the ball today at all am I? Indeed, transcoding would completely neglect the point of the original post. It brings my mind around to an interesting question though… If you're using a codec on your softphone that the provider does not support, with a reinvite switch codecs, drop the call, or just simply not reinvite? Alex *___________________________________________* Alex Robar, Technical Support, GearyTech Inc. 3075 Fourteenth Avenue, Unit 3, Markham, Ontario L3R 0G9 Markham: 905-513-8000 x 223 Fax: 905-513-8040 Toronto: 416-226-3614 Toll Free: 888-890-3499 [EMAIL PROTECTED] www.gearytech.com Strategic management of technology for business. Have you seen our new Voice over IP telephone systems for business? Visit http://www.gearytech.com/VoIP.asp for more information. ------------------------------ *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Ian Service *Sent:* Thursday, July 13, 2006 8:51 AM *Cc:* [email protected] *Subject:* Re: [on-asterisk] How to reduce bandwidth usage... ...and you can't transcode if you're trying to get out of the loop. - Ian On 7/13/06, *Alex Robar* <[EMAIL PROTECTED] > wrote: Ah, you're quite correct about that. Without transcoding you wouldn't be able to lower your bandwidth using free softphones. Alex *___________________________________________* Alex Robar , Technical Support, GearyTech Inc. 3075 Fourteenth Avenue, Unit 3, Markham, Ontario L3R 0G9 Markham : 905-513-8000 x 223 Fax: 905-513-8040 Toronto : 416-226-3614 Toll Free: 888-890-3499 [EMAIL PROTECTED] www.gearytech.com Strategic management of technology for business. Have you seen our new Voice over IP telephone systems for business? Visit http://www.gearytech.com/VoIP.asp for more information. ------------------------------ *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Ian Service *Sent:* Thursday, July 13, 2006 8:47 AM *Cc:* [email protected] *Subject:* Re: [on-asterisk] How to reduce bandwidth usage... Yep, but as far as I know Unlimitel only supports g711 and g729 so the only matching codec is g711. - Ian On 7/13/06, *Alex Robar* < [EMAIL PROTECTED]> wrote: Ian, > Also, you'll find that if you're using free soft phones, you're not going to be able to do anything other than g711 to the home users. The free Idefisk IAX softphone works great for us, and supports G.711, GSM, Speex and iLBC (no G.729 though). Alex *___________________________________________* Alex Robar , Technical Support, GearyTech Inc. 3075 Fourteenth Avenue, Unit 3, Markham, Ontario L3R 0G9 Markham : 905-513-8000 x 223 Fax: 905-513-8040 Toronto : 416-226-3614 Toll Free: 888-890-3499 [EMAIL PROTECTED] www.gearytech.com Strategic management of technology for business. Have you seen our new Voice over IP telephone systems for business? Visit http://www.gearytech.com/VoIP.asp for more information. ------------------------------ *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Ian Service *Sent:* Thursday, July 13, 2006 8:27 AM *To:* Chuck Mariotti *Cc:* [email protected] *Subject:* Re: [on-asterisk] How to reduce bandwidth usage... Hi Chuck, You're going to want to look into the reinvite configuration option of asterisk. Basically what happens is that your asterisk box receives the call then passes it off to the home user's internet connection and then gets out of the loop. There are a couple of issues of doing it this way though. The first is that if the users are using NAT to get their internet connection, your call may not be able to route correctly (you'll have to play with some connections to see what works and what doesn't). The second is that every time I've tried it, once your asterisk box drops out of the call, it doesn't seem to log any more call details. That could just be another setting that I haven't got set correctly though. Bottom line; it's quite doable without spending any extra and saving your precious bandwidth. Also, you'll find that if you're using free soft phones, you're not going to be able to do anything other than g711 to the home users. - Ian On 7/13/06, *Chuck Mariotti* < [EMAIL PROTECTED]> wrote: I am trying to figure out a way to reduce bandwidth usage for my Asterisk box which using Unlimitel is IAX2 trunks. I am trying to get remote sales staff (with highspeed connections) to use softphones (not finalized on which one to use). I would like all calls to go through Asterisk, but at the same time, reduce bandwidth usage for those incoming calls that go to these remote extensions at people's homes. How can I get it so that those home users get calls to their extensions, without using two active connections on my Asterisk box (taking up bandwidth)? As well, how can I get these same home users to make outbound calls, without taking two active connections up? I guess what I am asking is, how do I get the home users to be an extension that I can make calls and receive calls with a simple talk straight to unlimitel without talking too much to my asterisk box. I would like to capture and report on these extension on my Asterisk box (inbound/outbound calls), but could live without if I had to. I hope I'm just not getting a simple concept here and it already does this automatically. Regards, Chuck
