Re: [asterisk-biz] activex softphone

We quit using IAX years ago.

----- Original Message -----
From: "Steve Totaro" <stotaro@totarotechnologies.com>
To: "Commercial and Business-Oriented Asterisk Discussion"
<asterisk-biz@lists.digium.com>
Sent: Thursday, March 27, 2008 4:55 PM
Subject: Re: [asterisk-biz] activex softphone

>
> The army is the people that figure it out on their own or hire me to
> consult their "audio issues". Peek around, check for other
> possibilities and then switch them from IAX to SIP and, boom, no more
> audio issues.
>
> Not sure what qualifies an army but does hundreds count?
>
> Thanks,
> Steve Totaro
>
> On Thu, Mar 27, 2008 at 3:19 PM, Dean Collins <Dean@cognation.net> wrote:
>> Says you and who's army steve :)
>>
>>
>>
>> Regards,
>>
>> Dean Collins
>> Cognation Pty Ltd
>> dean@cognation.net
>> +1-212-203-4357
>> +61-2-9016-5642 (Sydney in-dial).
>>
>>
>>
>> > -----Original Message-----
>> > From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-
>> > bounces@lists.digium.com] On Behalf Of Steve Totaro
>> > Sent: Thursday, 27 March 2008 2:38 PM
>> > To: Commercial and Business-Oriented Asterisk Discussion
>>
>>
>> > Subject: Re: [asterisk-biz] activex softphone
>> >
>> > On Thu, Mar 27, 2008 at 1:13 PM, Tim Panton <thp@westhawk.co.uk>
>> wrote:
>> > >
>> > > On 21 Mar 2008, at 10:38, Andor Czafik (Akakiko) wrote:
>> > >
>> > > > Hi!
>> > > >
>> > > > I need control (answer, and call) sip phone from web browser, and
>> the
>> > > > best is, (i think) the activex softphone.
>> > > > What is the best, and cheaper(or free) activex softphone?
>> > > > Thanks
>> > > > Andor
>> > > >
>> > > > ________________
>> > >
>> > > Does it have to be SIP ? Would IAX do ?
>> > > Do you care about the other 25% of users who use Firefox or Macs?
>> > > Will your users agree to install an activeX control ?
>> > >
>> > > Tim.
>> > >
>> > >

www.phonefromhere.com
>> > >
>> >
>> >
>> > IAX is junk. Never figured out what causes the quality to be so poor,
>> > maybe it is simultaneous usage, maybe it is trunking enabled, maybe it
>> > is just junk.
>> >
>> > It seems to work OK for single calls to boxes taking single calls but
>> > beyond that, the audio is so bad it is not even worth the time trying
>> > to figure it out.
>> >
>> > SIP still rules.
>> >
>> > Thanks,
>> > Steve Totaro
>> >
>>
>>
>> > _______________________________________________
>> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
>> >
>> > asterisk-biz mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >

http://lists.digium.com/mailman/listinfo/asterisk-biz
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-biz mailing list
>> To UNSUBSCRIBE or update options visit:
>>

http://lists.digium.com/mailman/listinfo/asterisk-biz
>>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
>

http://lists.digium.com/mailman/listinfo/asterisk-biz

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-biz