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

Wed Mar 17 00:34:57 MST 2004

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
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at 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
> 