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:[EMAIL PROTECTED]:5060;transport=udp>
From: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060>;tag=014ee749 To: <sip:[EMAIL PROTECTED]:5060;transport=udp>;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]:5060>;tag=014ee749 To: <sip:[EMAIL PROTECTED]:5060;transport=udp>;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]:5060;transport=udp>
From: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060>;tag=6c3de14a To: <sip:[EMAIL PROTECTED]:5060;transport=udp>;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]:5060>;tag=6c3de14a To: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;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
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