[Asterisk-Users] New Firefly Beta - with SIP and G.729

[Adam Hart]

I'll look at it tomorrow, what url are you using? standard asterisk syntax?

Stig Andersson wrote:

>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 at asterisk.ymex.com:5060;transport=udp>
>From: <sip:stig at 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 at 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 at 217.119.162.35:5060>;tag=014ee749
>To: <sip:stig at 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 at 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 at 217.119.162.35:5060>;tag=014ee749
>To: <sip:stig at 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 at 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 at asterisk.ymex.com:5060;transport=udp>
>From: <sip:stig at 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 at 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 at 217.119.162.35:5060>;tag=6c3de14a
>To: <sip:stig at 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 at 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 at 217.119.162.35:5060>;tag=6c3de14a
>To: <sip:stig at 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 at 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
>>>      
>>>
>>_______________________________________________
>>    
>>