[Asterisk-Users] Re: FireFly w/ SIP

[Deon Rodden]

FireFly is awesome, it's not giving quality issues like X-Lite is. FireFly's
only problem was it wasn't registering with the server often enough, making
that NAT box forget the connection and not allow incoming streams. 

Adam Hart said they would add it as an adjustable feature to the next
version coming out next week. 


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Leah Newmark
Sent: Friday, October 15, 2004 9:28 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Re: FireFly w/ SIP

I can tell you that you are not alone. It's an issue I believe with Firefly,

and not in your configurations.

>
> Message: 8
> Date: Fri, 15 Oct 2004 13:06:17 +0200
> From: Willem de Groot <willem at byte.nl>
> Subject: [Asterisk-Users] FireFly w/ SIP
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 	<asterisk-users at lists.digium.com>
> Message-ID: <416FAF29.7070803 at byte.nl>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk?
>
> It works in IAX mode, but in SIP mode I am unable to hear anything (no
> dialtone, no voice). I am able to setup a conversation with another SIP
> phone though (Xlite, Grandstream) and the other side can hear the
> FireFly user just fine (both sides using g711u).
>
> I tried different PC's with different audio hardware. They all work fine
> using FireFly in IAX mode and using other softphones, so I guess it must
> be related so FireFly in SIP mode.
>
> This is my SIP config:
>
> [201]
> type=friend
> host=dynamic
> dtmfmode=rfc2833
> context=sip
> canreinvite=yes
>
> FireFly is also configured for rfc2833 dtmf.
>
> Thanks for any suggestions!
> Willem
>
>
>

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