As Promised, I've released a new version of Firefly (ver 1.8) with IAX & SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user -> software -> firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam
Adam Hart wrote:> As Promised, I've released a new version of Firefly (ver 1.8) with IAX & > SIP support back in.STUN support doesn't seem to work... Keeps saying unable to contact stun server, and when I did a packet dump and closed and reopened the prog several times I couldn't see any attempts to hit the stun server... STUN server in question (stun.e164.org) works fine with the BT101's...> If it crashes on startup, export your Firefly tree from the registry > (current user -> software -> firefly), then delete tree from your > registry. If that fixes it, send me your exported reg file, there's a > bug left to do with some wierd reg entry but everyone just deletes it > instead of sending it to me :|I freshly reinstalled my laptop over the weekend and haven't resinstalled firefly till now... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers
Lol, remove the 'r' from the url. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Adam Hart Sent: Monday, 31 May 2004 3:33 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] New Firefly version Just released a minor update http://www.virbiage.com/firefly/download/firefly-thirdparty.exer Fixed STUN - my code was for the old version of STUN RFC. Thanks to Duane for helping debug it. if port 5060 (sip) is in use, it doesn't crash on startup now - just an error message :) I'm guessing this has been a cause of many crashes, people having Xten running in the background. Thanks to Karl for the dump file on that one. keep the bugs coming, Adam PS hope you're enjoying the new contact groups :) Adam Hart wrote:> Duane wrote: > >> Adam Hart wrote: >> >>> As Promised, I've released a new version of Firefly (ver 1.8) with >>> IAX & SIP support back in. >> >> >> >> STUN support doesn't seem to work... Keeps saying unable to contact >> stun server, and when I did a packet dump and closed and reopened the>> prog several times I couldn't see any attempts to hit the stunserver...>> >> STUN server in question (stun.e164.org) works fine with theBT101's...>> >>> If it crashes on startup, export your Firefly tree from the registry>>> (current user -> software -> firefly), then delete tree from your >>> registry. If that fixes it, send me your exported reg file, there'sa>>> bug left to do with some wierd reg entry but everyone just deletesit>>> instead of sending it to me :| >> >> >> >> I freshly reinstalled my laptop over the weekend and haven't >> resinstalled firefly till now... >> > Oops, using a default stun port of 10000 - fixing now :) > _______________________________________________ > 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 >_______________________________________________ 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
Thanks Adam, no crash after installing over 1.5 B3388. However changing the SIP RTP Port is still not accepted. jo Adam Hart wrote:> As Promised, I've released a new version of Firefly (ver 1.8) with IAX > & SIP support back in. > > Get it from Virbiage site or here's the direct link > http://www.virbiage.com/firefly/download/firefly-thirdparty.exe > > If it crashes on startup, export your Firefly tree from the registry > (current user -> software -> firefly), then delete tree from your > registry. If that fixes it, send me your exported reg file, there's a > bug left to do with some wierd reg entry but everyone just deletes it > instead of sending it to me :| > > Transfers will be in the next version - email me any comments, > requested features, bugs and I'll see what I can do > > -Adam > _______________________________________________ > 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 >
Hello Adam, Hi Adam, two features I would really like to have: - the textbox from "Dial a URL" in the normal client (maybe optionally) so that you could easily copy and paste numbers in - a function that replaces +49 or wathever to 00. maybe it would be also possible, to recognize that +49 (333) 9999 is not a local number, so that another 0 should be added (or a 9). Regards Felix> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Adam Hart > Sent: Monday, May 31, 2004 3:01 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] New Firefly version > > As Promised, I've released a new version of Firefly (ver 1.8) > with IAX & SIP support back in. > > Get it from Virbiage site or here's the direct link > http://www.virbiage.com/firefly/download/firefly-thirdparty.exe > > If it crashes on startup, export your Firefly tree from the > registry (current user -> software -> firefly), then delete > tree from your registry. If that fixes it, send me your > exported reg file, there's a bug left to do with some wierd > reg entry but everyone just deletes it instead of sending it to me :| > > Transfers will be in the next version - email me any > comments, requested features, bugs and I'll see what I can do > > -Adam > _______________________________________________ > 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 >
Why all the time the firefly show me the message: Sip registration failed for the network Home (407). The server, username and password are correct. I'm using the default RTP port 5000 in the SIP tab. Using the SJPhone I can register; using the firefly I can call any registered number, but I can't register. On asterisk console: Sip read: REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 To: <sip:2003@192.168.199.3:5060;transport=udp> From: <sip:2003@192.168.199.121:5060>;tag=5a1c4f36 Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER Contact: <sip:2003@192.168.199.121:5060> Expires: 3600 Max-Forwards: 70 User-Agent: Firefly Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport From: <sip:2003@192.168.199.121:5060>;tag=5a1c4f36 To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2003@192.168.199.4> Content-Length: 0 to 192.168.199.121:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport From: <sip:2003@192.168.199.121:5060>;tag=5a1c4f36 To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2003@192.168.199.4> Proxy-Authenticate: Digest realm="asterisk", nonce="38165263" Content-Length: 0 to 192.168.199.121:5060 SAMPLANET1*CLI> Sip read: REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 To: <sip:2003@192.168.199.3:5060;transport=udp> From: <sip:2003@192.168.199.121:5060>;tag=6c3de14a Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER Contact: <sip:2003@192.168.199.121:5060> Expires: 3600 Max-Forwards: 70 Proxy-Authorization: Digest username=2003,realm="asterisk",nonce="38165263",uri="sip:192.168.199.3:5060; transport=udp",response="ec0afc0a2b13a725aa40b5c311c396d8",algorithm=MD5 User-Agent: Firefly Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: <sip:2003@192.168.199.121:5060>;tag=6c3de14a To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2003@192.168.199.4> Content-Length: 0 to 192.168.199.121:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: <sip:2003@192.168.199.121:5060>;tag=6c3de14a To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2003@192.168.199.4> Proxy-Authenticate: Digest realm="asterisk", nonce="38165263" Content-Length: 0 to 192.168.199.121:5060
An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040602/35d3eca2/attachment.htm -------------- next part -------------- ? u hv to change ur sip.conf & extensions.conf file if u want i will send u. On Tue, 01 Jun 2004 Paul Mahler wrote :>I'm having this problem too. > > >Paul Mahler >pmahler@signate.com >Signate, LLC >665 Third Street >Suite 100 >San Francisco, CA > 94107-1901 > > Asterisk Services and Training > > > > > > > > > > > -----Original Message----- > > From: asterisk-users-admin@lists.digium.com > > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > > miguel@amplanet.com.br > > Sent: Tuesday, June 01, 2004 7:53 AM > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] New Firefly version > > > > Why all the time the firefly show me the message: Sip > > registration failed for the network Home (407). > > > > The server, username and password are correct. I'm using the > > default RTP port 5000 in the SIP tab. > > > > Using the SJPhone I can register; using the firefly I can > > call any registered number, but I can't register. > > > > On asterisk console: > > > > Sip read: > > REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 > > To: <sip:2003@192.168.199.3:5060;transport=udp> > > From: <sip:2003@192.168.199.121:5060>;tag=5a1c4f36 > > Via: SIP/2.0/UDP > > 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport > > Call-ID: c90fa011e82acf3e > > CSeq: 1 REGISTER > > Contact: <sip:2003@192.168.199.121:5060> > > Expires: 3600 > > Max-Forwards: 70 > > User-Agent: Firefly > > Content-Length: 0 > > > > > > 11 headers, 0 lines > > Using latest request as basis request > > Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP > > 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport > > From: <sip:2003@192.168.199.121:5060>;tag=5a1c4f36 > > To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 > > Call-ID: c90fa011e82acf3e > > CSeq: 1 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Contact: <sip:2003@192.168.199.4> > > Content-Length: 0 > > > > > > to 192.168.199.121:5060 > > Transmitting (no NAT): > > SIP/2.0 407 Proxy Authentication Required > > Via: SIP/2.0/UDP > > 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport > > From: <sip:2003@192.168.199.121:5060>;tag=5a1c4f36 > > To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 > > Call-ID: c90fa011e82acf3e > > CSeq: 1 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Contact: <sip:2003@192.168.199.4> > > Proxy-Authenticate: Digest realm="asterisk", nonce="38165263" > > Content-Length: 0 > > > > > > to 192.168.199.121:5060 > > SAMPLANET1*CLI> > > > > Sip read: > > REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 > > To: <sip:2003@192.168.199.3:5060;transport=udp> > > From: <sip:2003@192.168.199.121:5060>;tag=6c3de14a > > Via: SIP/2.0/UDP > > 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport > > Call-ID: c90fa011e82acf3e > > CSeq: 1 REGISTER > > Contact: <sip:2003@192.168.199.121:5060> > > Expires: 3600 > > Max-Forwards: 70 > > Proxy-Authorization: Digest > > username=2003,realm="asterisk",nonce="38165263",uri="sip:192.1 > > 68.199.3:5060; > > transport=udp",response="ec0afc0a2b13a725aa40b5c311c396d8",alg > > orithm=MD5 > > User-Agent: Firefly > > Content-Length: 0 > > > > > > 12 headers, 0 lines > > Using latest request as basis request > > Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP > > 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport > > From: <sip:2003@192.168.199.121:5060>;tag=6c3de14a > > To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 > > Call-ID: c90fa011e82acf3e > > CSeq: 1 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Contact: <sip:2003@192.168.199.4> > > Content-Length: 0 > > > > > > to 192.168.199.121:5060 > > Transmitting (no NAT): > > SIP/2.0 407 Proxy Authentication Required > > Via: SIP/2.0/UDP > > 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport > > From: <sip:2003@192.168.199.121:5060>;tag=6c3de14a > > To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 > > Call-ID: c90fa011e82acf3e > > CSeq: 1 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > > Contact: <sip:2003@192.168.199.4> > > Proxy-Authenticate: Digest realm="asterisk", nonce="38165263" > > Content-Length: 0 > > > > > > to 192.168.199.121:5060 > > > > > > _______________________________________________ > > 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 > > > >_______________________________________________ >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
There's a new version out with some bugs fixed major ones fixed: deadlock on call end, iax thread getting locked out, few contact group list bugs, one on exit crash bug fixed I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe -Adam Adam Hart wrote:> As Promised, I've released a new version of Firefly (ver 1.8) with IAX & > SIP support back in. > > Get it from Virbiage site or here's the direct link > http://www.virbiage.com/firefly/download/firefly-thirdparty.exe > > If it crashes on startup, export your Firefly tree from the registry > (current user -> software -> firefly), then delete tree from your > registry. If that fixes it, send me your exported reg file, there's a > bug left to do with some wierd reg entry but everyone just deletes it > instead of sending it to me :| > > Transfers will be in the next version - email me any comments, requested > features, bugs and I'll see what I can do > > -Adam > _______________________________________________ > 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 >
I have this two ip at the same machine, but I tried it using the both address, the result is the same. Kind regards, Miguel Date: Wed, 02 Jun 2004 13:50:05 +1000 From: Adam Hart <adam@teragen.com.au> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] New Firefly version Reply-To: asterisk-users@lists.digium.com the log looks legit except why does asterisk have a different IP in the contact compared to the 'to' address. I can connect successfully to my asterisk server and FWD - can anyone give me sip access to a asterisk server that firefly doesn't work on? miguel@amplanet.com.br wrote:> Why all the time the firefly show me the message: Sip registration > failed for the network Home (407). > > The server, username and password are correct. I'm using the default > RTP port 5000 in the SIP tab. > > Using the SJPhone I can register; using the firefly I can call any > registered number, but I can't register. > > On asterisk console: > > Sip read: > REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 > To: <sip:2003@192.168.199.3:5060;transport=udp> > From: <sip:2003@192.168.199.121:5060>;tag=5a1c4f36 > Via: SIP/2.0/UDP > 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport > Call-ID: c90fa011e82acf3e > CSeq: 1 REGISTER > Contact: <sip:2003@192.168.199.121:5060> > Expires: 3600 > Max-Forwards: 70 > User-Agent: Firefly > Content-Length: 0 > > > 11 headers, 0 lines > Using latest request as basis request > Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport > From: <sip:2003@192.168.199.121:5060>;tag=5a1c4f36 > To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 > Call-ID: c90fa011e82acf3e > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:2003@192.168.199.4> > Content-Length: 0 > > > to 192.168.199.121:5060 > Transmitting (no NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport > From: <sip:2003@192.168.199.121:5060>;tag=5a1c4f36 > To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 > Call-ID: c90fa011e82acf3e > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:2003@192.168.199.4> > Proxy-Authenticate: Digest realm="asterisk", nonce="38165263" > Content-Length: 0 > > > to 192.168.199.121:5060 > SAMPLANET1*CLI> > > Sip read: > REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 > To: <sip:2003@192.168.199.3:5060;transport=udp> > From: <sip:2003@192.168.199.121:5060>;tag=6c3de14a > Via: SIP/2.0/UDP > 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport > Call-ID: c90fa011e82acf3e > CSeq: 1 REGISTER > Contact: <sip:2003@192.168.199.121:5060> > Expires: 3600 > Max-Forwards: 70 > Proxy-Authorization: Digest > username=2003,realm="asterisk",nonce="38165263",uri="sip:192.168.199.3 > :5060; > transport=udp",response="ec0afc0a2b13a725aa40b5c311c396d8",algorithm=M > D5 > User-Agent: Firefly > Content-Length: 0 > > > 12 headers, 0 lines > Using latest request as basis request > Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport > From: <sip:2003@192.168.199.121:5060>;tag=6c3de14a > To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 > Call-ID: c90fa011e82acf3e > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:2003@192.168.199.4> > Content-Length: 0 > > > to 192.168.199.121:5060 > Transmitting (no NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport > From: <sip:2003@192.168.199.121:5060>;tag=6c3de14a > To: <sip:2003@192.168.199.3:5060;transport=udp>;tag=as7843e908 > Call-ID: c90fa011e82acf3e > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:2003@192.168.199.4> > Proxy-Authenticate: Digest realm="asterisk", nonce="38165263" > Content-Length: 0 > > > to 192.168.199.121:5060 > > > _______________________________________________ > 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 >
Please, send to me. Kind regards, Miguel Date: 2 Jun 2004 04:39:39 -0000 From: "muralikrishnan lakshmanan" <slmurali@rediffmail.com> To: asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] New Firefly version Reply-To: asterisk-users@lists.digium.com This is a multipart mime message --Next_1086151179---0-202.54.124.130-19795 Content-type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable Content-Disposition: inline <P>=0A <BR>=0Au hv to change ur sip.conf & extensions.conf file if u want i will send u.<BR>=0A<BR>=0A<BR>=0AOn Tue, 01 Jun 2004 Paul Mahler wrote :<BR>=0A>I'm having this problem too.<BR>=0A><BR>=0A><BR>=0A>Paul Mahler<BR>=0A>pmahler@signate.com<BR>=0A>Signate, LLC<BR>=0A>665 Third Street<BR>=0A>Suite 100<BR>=0A>San Francisco, CA<BR>=0A> 94107-1901<BR>=0A><BR>=0A> Asterisk Services and Training<BR>=0A><BR>=0A><BR>=0A><BR>=0A><BR>=0A><BR>=0A><BR>=0A><BR>=0A><BR>=0A><BR>=0A> > -----Original Message-----<BR>=0A> > From: asterisk-users-admin@lists.digium.com<BR>=0A> > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of<BR>=0A> > miguel@amplanet.com.br<BR>=0A> > Sent: Tuesday, June 01, 2004 7:53 AM<BR>=0A> > To: asterisk-users@lists.digium.com<BR>=0A> > Subject: [Asterisk-Users] New Firefly version<BR>=0A> ><BR>=0A> > Why all the time the firefly show me the message: Sip<BR>=0A> > registration failed for the network Home (407).<BR>=0A> ><BR>=0A> > The server, username and password are correct. I'm using the<BR>=0A> > default RTP port 5000 in the SIP tab.<BR>=0A> ><BR>=0A> > Using the SJPhone I can register; using the firefly I can<BR>=0A> > call any registered number, but I can't register.<BR>=0A> ><BR>=0A> > On asterisk console:<BR>=0A> ><BR>=0A> > Sip read:<BR>=0A> > REGISTER sip:192.168.199.3:5060;transport=3Dudp SIP/2.0<BR>=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp><BR>=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D5a1c4f36<BR>=0A> > Via: SIP/2.0/UDP<BR>=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-436999556-1--c87542-;rport<BR>=0A> > Call-ID: c90fa011e82acf3e<BR>=0A> > CSeq: 1 REGISTER<BR>=0A> > Contact: <sip:2003@192.168.199.121:5060><BR>=0A> > Expires: 3600<BR>=0A> > Max-Forwards: 70<BR>=0A> > User-Agent: Firefly<BR>=0A> > Content-Length: 0<BR>=0A> ><BR>=0A> ><BR>=0A> > 11 headers, 0 lines<BR>=0A> > Using latest request as basis request<BR>=0A> > Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT):<BR>=0A> > SIP/2.0 100 Trying<BR>=0A> > Via: SIP/2.0/UDP<BR>=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-436999556-1--c87542-;rport<BR>=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D5a1c4f36<BR>=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp>;tag=3Das7843e908<BR>=0A> > Call-ID: c90fa011e82acf3e<BR>=0A> > CSeq: 1 REGISTER<BR>=0A> > User-Agent: Asterisk PBX<BR>=0A> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>=0A> > Contact: <sip:2003@192.168.199.4><BR>=0A> > Content-Length: 0<BR>=0A> ><BR>=0A> ><BR>=0A> > to 192.168.199.121:5060<BR>=0A> > Transmitting (no NAT):<BR>=0A> > SIP/2.0 407 Proxy Authentication Required<BR>=0A> > Via: SIP/2.0/UDP<BR>=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-436999556-1--c87542-;rport<BR>=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D5a1c4f36<BR>=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp>;tag=3Das7843e908<BR>=0A> > Call-ID: c90fa011e82acf3e<BR>=0A> > CSeq: 1 REGISTER<BR>=0A> > User-Agent: Asterisk PBX<BR>=0A> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>=0A> > Contact: <sip:2003@192.168.199.4><BR>=0A> > Proxy-Authenticate: Digest realm=3D"asterisk", nonce=3D"38165263"<BR>=0A> > Content-Length: 0<BR>=0A> ><BR>=0A> ><BR>=0A> > to 192.168.199.121:5060<BR>=0A> > SAMPLANET1*CLI><BR>=0A> ><BR>=0A> > Sip read:<BR>=0A> > REGISTER sip:192.168.199.3:5060;transport=3Dudp SIP/2.0<BR>=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp><BR>=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D6c3de14a<BR>=0A> > Via: SIP/2.0/UDP<BR>=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-373911025-1--c87542-;rport<BR>=0A> > Call-ID: c90fa011e82acf3e<BR>=0A> > CSeq: 1 REGISTER<BR>=0A> > Contact: <sip:2003@192.168.199.121:5060><BR>=0A> > Expires: 3600<BR>=0A> > Max-Forwards: 70<BR>=0A> > Proxy-Authorization: Digest<BR>=0A> > username=3D2003,realm=3D"asterisk",nonce=3D"38165263",uri=3D"sip:192.1<BR>=0A> > 68.199.3:5060;<BR>=0A> > transport=3Dudp",response=3D"ec0afc0a2b13a725aa40b5c311c396d8",alg<BR>=0A> > orithm=3DMD5<BR>=0A> > User-Agent: Firefly<BR>=0A> > Content-Length: 0<BR>=0A> ><BR>=0A> ><BR>=0A> > 12 headers, 0 lines<BR>=0A> > Using latest request as basis request<BR>=0A> > Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT):<BR>=0A> > SIP/2.0 100 Trying<BR>=0A> > Via: SIP/2.0/UDP<BR>=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-373911025-1--c87542-;rport<BR>=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D6c3de14a<BR>=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp>;tag=3Das7843e908<BR>=0A> > Call-ID: c90fa011e82acf3e<BR>=0A> > CSeq: 1 REGISTER<BR>=0A> > User-Agent: Asterisk PBX<BR>=0A> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>=0A> > Contact: <sip:2003@192.168.199.4><BR>=0A> > Content-Length: 0<BR>=0A> ><BR>=0A> ><BR>=0A> > to 192.168.199.121:5060<BR>=0A> > Transmitting (no NAT):<BR>=0A> > SIP/2.0 407 Proxy Authentication Required<BR>=0A> > Via: SIP/2.0/UDP<BR>=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-373911025-1--c87542-;rport<BR>=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D6c3de14a<BR>=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp>;tag=3Das7843e908<BR>=0A> > Call-ID: c90fa011e82acf3e<BR>=0A> > CSeq: 1 REGISTER<BR>=0A> > User-Agent: Asterisk PBX<BR>=0A> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<BR>=0A> > Contact: <sip:2003@192.168.199.4><BR>=0A> > Proxy-Authenticate: Digest realm=3D"asterisk", nonce=3D"38165263"<BR>=0A> > Content-Length: 0<BR>=0A> ><BR>=0A> ><BR>=0A> > to 192.168.199.121:5060<BR>=0A> ><BR>=0A> ><BR>=0A> > _______________________________________________<BR>=0A> > Asterisk-Users mailing list<BR>=0A> > Asterisk-Users@lists.digium.com<BR>=0A> > http://lists.digium.com/mailman/listinfo/asterisk-users<BR>=0A> > To UNSUBSCRIBE or update options visit:<BR>=0A> > http://lists.digium.com/mailman/listinfo/asterisk-users<BR>=0A> ><BR>=0A><BR>=0A>_______________________________________________<BR>=0A>Asterisk-Users mailing list<BR>=0A>Asterisk-Users@lists.digium.com<BR>=0A>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>=0A>To UNSUBSCRIBE or update options visit:<BR>=0A> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>=0A=0A</P>=0A=0A=0A<br><br>=0A<A target=3D"_blank" HREF=3D"http://clients.rediff.com/signature/track_sig.asp"><IMG SRC=3D"http://ads.rediff.com/RealMedia/ads/adstream_nx.cgi/www.rediffmail.com/inbox.htm@Bottom" BORDER=3D0 VSPACE=3D0 HSPACE=3D0></a>=0A --Next_1086151179---0-202.54.124.130-19795 Content-type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable Content-Disposition: inline =A0=0Au hv to change ur sip.conf & extensions.conf file if u want i will send u.=0A=0A=0AOn Tue, 01 Jun 2004 Paul Mahler wrote :=0A>I'm having this problem too.=0A>=0A>=0A>Paul Mahler=0A>pmahler@signate.com=0A>Signate, LLC=0A>665 Third Street=0A>Suite 100=0A>San Francisco, CA=0A> 94107-1901=0A>=0A> Asterisk Services and Training=0A>=0A>=0A>=0A>=0A>=0A>=0A>=0A>=0A>=0A> > -----Original Message-----=0A> > From: asterisk-users-admin@lists.digium.com=0A> > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of=0A> > miguel@amplanet.com.br=0A> > Sent: Tuesday, June 01, 2004 7:53 AM=0A> > To: asterisk-users@lists.digium.com=0A> > Subject: [Asterisk-Users] New Firefly version=0A> >=0A> > Why all the time the firefly show me the message: Sip=0A> > registration failed for the network Home (407).=0A> >=0A> > The server, username and password are correct. I'm using the=0A> > default RTP port 5000 in the SIP tab.=0A> >=0A> > Using the SJPhone I can register; using the firefly I can=0A> > call any registered number, but I can't register.=0A> >=0A> > On asterisk console:=0A> >=0A> > Sip read:=0A> > REGISTER sip:192.168.199.3:5060;transport=3Dudp SIP/2.0=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp>=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D5a1c4f36=0A> > Via: SIP/2.0/UDP=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-436999556-1--c87542-;rport=0A> > Call-ID: c90fa011e82acf3e=0A> > CSeq: 1 REGISTER=0A> > Contact: <sip:2003@192.168.199.121:5060>=0A> > Expires: 3600=0A> > Max-Forwards: 70=0A> > User-Agent: Firefly=0A> > Content-Length: 0=0A> >=0A> >=0A> > 11 headers, 0 lines=0A> > Using latest request as basis request=0A> > Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT):=0A> > SIP/2.0 100 Trying=0A> > Via: SIP/2.0/UDP=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-436999556-1--c87542-;rport=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D5a1c4f36=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp>;tag=3Das7843e908=0A> > Call-ID: c90fa011e82acf3e=0A> > CSeq: 1 REGISTER=0A> > User-Agent: Asterisk PBX=0A> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER=0A> > Contact: <sip:2003@192.168.199.4>=0A> > Content-Length: 0=0A> >=0A> >=0A> > to 192.168.199.121:5060=0A> > Transmitting (no NAT):=0A> > SIP/2.0 407 Proxy Authentication Required=0A> > Via: SIP/2.0/UDP=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-436999556-1--c87542-;rport=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D5a1c4f36=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp>;tag=3Das7843e908=0A> > Call-ID: c90fa011e82acf3e=0A> > CSeq: 1 REGISTER=0A> > User-Agent: Asterisk PBX=0A> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER=0A> > Contact: <sip:2003@192.168.199.4>=0A> > Proxy-Authenticate: Digest realm=3D"asterisk", nonce=3D"38165263"=0A> > Content-Length: 0=0A> >=0A> >=0A> > to 192.168.199.121:5060=0A> > SAMPLANET1*CLI>=0A> >=0A> > Sip read:=0A> > REGISTER sip:192.168.199.3:5060;transport=3Dudp SIP/2.0=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp>=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D6c3de14a=0A> > Via: SIP/2.0/UDP=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-373911025-1--c87542-;rport=0A> > Call-ID: c90fa011e82acf3e=0A> > CSeq: 1 REGISTER=0A> > Contact: <sip:2003@192.168.199.121:5060>=0A> > Expires: 3600=0A> > Max-Forwards: 70=0A> > Proxy-Authorization: Digest=0A> > username=3D2003,realm=3D"asterisk",nonce=3D"38165263",uri=3D"sip:192.1=0A> > 68.199.3:5060;=0A> > transport=3Dudp",response=3D"ec0afc0a2b13a725aa40b5c311c396d8",alg=0A> > orithm=3DMD5=0A> > User-Agent: Firefly=0A> > Content-Length: 0=0A> >=0A> >=0A> > 12 headers, 0 lines=0A> > Using latest request as basis request=0A> > Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT):=0A> > SIP/2.0 100 Trying=0A> > Via: SIP/2.0/UDP=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-373911025-1--c87542-;rport=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D6c3de14a=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp>;tag=3Das7843e908=0A> > Call-ID: c90fa011e82acf3e=0A> > CSeq: 1 REGISTER=0A> > User-Agent: Asterisk PBX=0A> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER=0A> > Contact: <sip:2003@192.168.199.4>=0A> > Content-Length: 0=0A> >=0A> >=0A> > to 192.168.199.121:5060=0A> > Transmitting (no NAT):=0A> > SIP/2.0 407 Proxy Authentication Required=0A> > Via: SIP/2.0/UDP=0A> > 192.168.199.121:5060;branch=3Dz9hG4bK-c87542-373911025-1--c87542-;rport=0A> > From: <sip:2003@192.168.199.121:5060>;tag=3D6c3de14a=0A> > To: <sip:2003@192.168.199.3:5060;transport=3Dudp>;tag=3Das7843e908=0A> > Call-ID: c90fa011e82acf3e=0A> > CSeq: 1 REGISTER=0A> > User-Agent: Asterisk PBX=0A> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER=0A> > Contact: <sip:2003@192.168.199.4>=0A> > Proxy-Authenticate: Digest realm=3D"asterisk", nonce=3D"38165263"=0A> > Content-Length: 0=0A> >=0A> >=0A> > to 192.168.199.121:5060=0A> >=0A> >=0A> > _______________________________________________=0A> > Asterisk-Users mailing list=0A> > Asterisk-Users@lists.digium.com=0A> > http://lists.digium.com/mailman/listinfo/asterisk-users=0A> > To UNSUBSCRIBE or update options visit:=0A> > http://lists.digium.com/mailman/listinfo/asterisk-users=0A> >=0A>=0A>_______________________________________________=0A>Asterisk-Users mailing list=0A>Asterisk-Users@lists.digium.com=0A>http://lists.digium.com/mailman/listinfo/asterisk-users=0A>To UNSUBSCRIBE or update options visit:=0A> http://lists.digium.com/mailman/listinfo/asterisk-users=0A=0A=0A --Next_1086151179---0-202.54.124.130-19795-- --__--__-- _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest
As always, I'm happy to announce a new version of Firefly. Firefly 1.9.8 has more of what you want and less of what you don't http://www.virbiage.com/firefly/download/firefly-thirdparty.exe There's a few bug fixes - notably fixed the Reject button and sending of audio before answering in some circumstances. -Adam
Also sound quality seems to be poor using the ULAW codec. I am using: - latest Firefly on Windows XP SP2 - Asterisk 1.0.5 patched coupled with Bristuff-0.2.0-RC5 with Florz patch for zaphfc - Linux kernel 2.6.9-1.681_FC3 Fedora Core 3 (obviously) - connecting to FWD dialing 411 info service Any other codec is better and useable. Clearly it seems to be optimized for iLBC. ULAW is unusable for me. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of hhandresen Sent: Thursday, January 27, 2005 11:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: New Firefly version Hi Adam, Sory to say it, bu it still interupt the mouse if you have microsoft wireless mouse/keayboard. The mouse jumps around on the screen. Any news on this ? /HHA Adam Hart wrote:> As always, I'm happy to announce a new version of Firefly. > > Firefly 1.9.8 has more of what you want and less of what you don't > > http://www.virbiage.com/firefly/download/firefly-thirdparty.exe > > There's a few bug fixes - notably fixed the Reject button and sending > of audio before answering in some circumstances. > > -Adam > _______________________________________________ > 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 >_______________________________________________ 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