I've been sitting on this release for a week so I thought I'd better
just release it :) Firefly now has SIP but it's still in a beta state.
If you manage to crash it, send me the hex address of the crash. If you
find it doesn't work with another SIP phone, let me know and I'll happy
get it working for you. I'll be interested to hear people's experiences
behind NATs.
To download the beta version of Firefly:
http://www.virbiage.com/firefly/download/firefly-dev.exe
(the current stable version of firefly will not have sip or g.729)
G729 support via dll - basically as we all know, G.729 ain't free but
you can get a free development version from Voiceage (Sipro), so I've
added support for using that. Download
http://www.virbiage.com/firefly/download/g729.zip and follow the
instructions in the Readme. You'll need to agree to their license and
download their library.
Firefly's Protocol Support now is:
Voip Protocols: SIP, IAX
Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL)
Next major feature will be conferencing.
feel free to email me,
Adam Hart
----- Original Message ----- From: "Adam Hart" To: <asterisk-users@lists.digium.com> Sent: Monday, March 15, 2004 6:32 PM Subject: [Asterisk-Users] New Firefly Beta - with SIP and G.729> Firefly's Protocol Support now is: > > Voip Protocols: SIP, IAX > Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL) >Sounds good. Any plans for Speex codec support?
Just a quick update, there's was a problem with SIP - if you were
getting SIP registration failed, grab the new version.
(http://www.virbiage.com/firefly/download/firefly-dev.exe)
thanks for the feedback about this bug,
Adam
Adam Hart wrote:
> I've been sitting on this release for a week so I thought I'd
better
> just release it :) Firefly now has SIP but it's still in a beta state.
> If you manage to crash it, send me the hex address of the crash. If
> you find it doesn't work with another SIP phone, let me know and
I'll
> happy get it working for you. I'll be interested to hear people's
> experiences behind NATs.
>
> To download the beta version of Firefly:
> http://www.virbiage.com/firefly/download/firefly-dev.exe
> (the current stable version of firefly will not have sip or g.729)
>
> G729 support via dll - basically as we all know, G.729 ain't free but
> you can get a free development version from Voiceage (Sipro), so I've
> added support for using that. Download
> http://www.virbiage.com/firefly/download/g729.zip and follow the
> instructions in the Readme. You'll need to agree to their license and
> download their library.
>
> Firefly's Protocol Support now is:
>
> Voip Protocols: SIP, IAX
> Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL)
>
> Next major feature will be conferencing.
>
> feel free to email me,
>
> Adam Hart
Stig Andersson
2004-Mar-17 00:34 UTC
[Asterisk-Users] New Firefly Beta - with SIP and G.729
Hi again, Installed your new release today (after the sip bugfix). Now SIP registers OK with asterisk, but calling fails... Firefly says: Couldn't start call. Asterisk in SIP debug mode shows the registration, but shows no response when firefly tries to call. Using NO stun, asterisk and Firefly on the same net, using only code G:711 u/alaw Registration data follows if of interrest... Regards Stig ------------- Sip read: REGISTER sip:asterisk.ymex.com:5060;transport=udp SIP/2.0 To: <sip:stig@asterisk.ymex.com:5060;transport=udp> From: <sip:stig@217.119.162.35:5060>;tag=014ee749 Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-386924776-1--c87542-;rport Call-ID: c75e00726c471711 CSeq: 1 REGISTER Contact: <sip:stig@217.119.162.35:5060> Expires: 3600 Max-Forwards: 70 User-Agent: Firefly Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 217.119.162.35 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-386924776-1--c87542-;rport From: <sip:stig@217.119.162.35:5060>;tag=014ee749 To: <sip:stig@asterisk.ymex.com:5060;transport=udp>;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:stig@217.119.162.48> Content-Length: 0 to 217.119.162.35:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-386924776-1--c87542-;rport From: <sip:stig@217.119.162.35:5060>;tag=014ee749 To: <sip:stig@asterisk.ymex.com:5060;transport=udp>;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:stig@217.119.162.48> Proxy-Authenticate: Digest realm="asterisk", nonce="30bc622a" Content-Length: 0 to 217.119.162.35:5060 asterisk*CLI> Sip read: REGISTER sip:asterisk.ymex.com:5060;transport=udp SIP/2.0 To: <sip:stig@asterisk.ymex.com:5060;transport=udp> From: <sip:stig@217.119.162.35:5060>;tag=6c3de14a Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport Call-ID: c75e00726c471711 CSeq: 1 REGISTER Contact: <sip:stig@217.119.162.35:5060> Expires: 3600 Max-Forwards: 70 Proxy-Authorization: Digest username=stig,realm="asterisk",nonce="30bc622a",uri="sip:asterisk.ymex.com:5060;transport=udp",response="d39488505ce4c15723e4b8f3a7a2bb69",algorithm=MD5 User-Agent: Firefly Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 217.119.162.35 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: <sip:stig@217.119.162.35:5060>;tag=6c3de14a To: <sip:stig@asterisk.ymex.com:5060;transport=udp>;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:stig@217.119.162.48> Content-Length: 0 to 217.119.162.35:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: <sip:stig@217.119.162.35:5060>;tag=6c3de14a To: <sip:stig@asterisk.ymex.com:5060;transport=udp>;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:stig@217.119.162.48>;expires=3600 Date: Wed, 17 Mar 2004 07:24:46 GMT Content-Length: 0 to 217.119.162.35:5060 At 17:34 2004-03-17 +1100, you wrote:>Just a quick update, there's was a problem with SIP - if you were >getting SIP registration failed, grab the new version. >(http://www.virbiage.com/firefly/download/firefly-dev.exe) > >thanks for the feedback about this bug, > > Adam > >Adam Hart wrote: > >> I've been sitting on this release for a week so I thought I'd better >> just release it :) Firefly now has SIP but it's still in a beta state. >> If you manage to crash it, send me the hex address of the crash. If >> you find it doesn't work with another SIP phone, let me know and I'll >> happy get it working for you. I'll be interested to hear people's >> experiences behind NATs. >> >> To download the beta version of Firefly: >> http://www.virbiage.com/firefly/download/firefly-dev.exe >> (the current stable version of firefly will not have sip or g.729) >> >> G729 support via dll - basically as we all know, G.729 ain't free but >> you can get a free development version from Voiceage (Sipro), so I've >> added support for using that. Download >> http://www.virbiage.com/firefly/download/g729.zip and follow the >> instructions in the Readme. You'll need to agree to their license and >> download their library. >> >> Firefly's Protocol Support now is: >> >> Voip Protocols: SIP, IAX >> Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL) >> >> Next major feature will be conferencing. >> >> feel free to email me, >> >> Adam Hart > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >