Re: [Asterisk-Users] New Firefly version
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. Has anyone been able to make firefly work under wine at all? If so how? A decent linux client is the only thing skype has over SIP/IAX... -- 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 I do not try to dance better than anyone else. I only try to dance better than myself. ___ 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
Re: [Asterisk-Users] New Firefly version
Duane wrote: 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. Has anyone been able to make firefly work under wine at all? If so how? A decent linux client is the only thing skype has over SIP/IAX... Few people have claimed success, I'm not sure how though. ___ 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
Re: [Asterisk-Users] New Firefly version
Adam Hart wrote: Few people have claimed success, I'm not sure how though. Any chance of a native linux version then? :) -- 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 In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ 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] New Firefly version
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
Re: [Asterisk-Users] New Firefly version
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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
On Wed, 2 Jun 2004, Adam Hart wrote: I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Can I recommend you label files with version numbering - this must be about the third ? fourth ? firefly-thirdparty you've released. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
true, it's internally versioned though - look at the build number. But yes, I'll start suffixing a buildnumber on the files. i'm hoping this will be the last release before the magic feature called conferencing, unless this sip registration issue is firefly related -Adam gARetH baBB wrote: On Wed, 2 Jun 2004, Adam Hart wrote: I'd highly recommend upgrading to it http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Can I recommend you label files with version numbering - this must be about the third ? fourth ? firefly-thirdparty you've released. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Adam The link doesn't seems to work. Get back the following: Parse error: parse error, unexpected T_STRING in /usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line 121 Reto 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
fixed Reto Stauss wrote: Adam The link doesn't seems to work. Get back the following: Parse error: parse error, unexpected T_STRING in /usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line 121 Reto 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Thanks, is working now. Reto Zitat von Adam Hart [EMAIL PROTECTED]: fixed Reto Stauss wrote: Adam The link doesn't seems to work. Get back the following: Parse error: parse error, unexpected T_STRING in /usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line 121 Reto 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
On Wed, Jun 02, 2004 at 08:14:44AM +0100, gARetH baBB wrote: On Wed, 2 Jun 2004, Adam Hart wrote: Can I recommend you label files with version numbering - this must be about the third ? fourth ? firefly-thirdparty you've released. .. but have firefly-thirdparty.exe be a symbolic link to the latest version? My 2p. Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
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 [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Firefly version Reply-To: [EMAIL PROTECTED] 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? [EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]: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:[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 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:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] New Firefly version
Please, send to me. Kind regards, Miguel Date: 2 Jun 2004 04:39:39 - From: muralikrishnan lakshmanan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: RE: [Asterisk-Users] New Firefly version Reply-To: [EMAIL PROTECTED] 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=0Anbsp; BR=0Au hv to change ur sip.conf amp; extensions.conf file i= f u want i will send u.BR=0ABR=0ABR=0AOn Tue, 01 Jun 2004 Paul Mahler= wrote :BR=0Agt;I'm having this problem too.BR=0Agt;BR=0Agt;BR= =0Agt;Paul MahlerBR=0Agt;[EMAIL PROTECTED]BR=0Agt;Signate, LLCBR= =0Agt;665 Third StreetBR=0Agt;Suite 100BR=0Agt;San Francisco, CABR= =0Agt;nbsp; 94107-1901BR=0Agt;BR=0Agt;nbsp; Asterisk Services and = TrainingBR=0Agt;BR=0Agt;BR=0Agt;BR=0Agt;BR=0Agt;BR=0Agt;= BR=0Agt;BR=0Agt;BR=0Agt;BR=0Agt; gt; -Original Message-= BR=0Agt; gt; From: [EMAIL PROTECTED]BR=0Agt; gt= ; [mailto:[EMAIL PROTECTED] On Behalf OfBR=0Agt; g= t; [EMAIL PROTECTED]BR=0Agt; gt; Sent: Tuesday, June 01, 2004 7:53= AMBR=0Agt; gt; To: [EMAIL PROTECTED]BR=0Agt; gt; Sub= ject: [Asterisk-Users] New Firefly versionBR=0Agt; gt;BR=0Agt; gt; = Why all the time the firefly show me the message: SipBR=0Agt; gt; regis= tration failed for the network Home (407).BR=0Agt; gt;BR=0Agt; gt; = The server, username and password are correct. I'm using theBR=0Agt; gt= ; default RTP port 5000 in the SIP tab.BR=0Agt; gt;BR=0Agt; gt; Usi= ng the SJPhone I can register; using the firefly I canBR=0Agt; gt; call= any registered number, but I can't register.BR=0Agt; gt;BR=0Agt; g= t; On asterisk console:BR=0Agt; gt;BR=0Agt; gt; Sip read:BR=0Agt= ; gt; REGISTER sip:192.168.199.3:5060;transport=3Dudp SIP/2.0BR=0Agt; = gt; To: lt;sip:[EMAIL PROTECTED]:5060;transport=3Dudpgt;BR=0Agt; gt;= From: lt;sip:[EMAIL PROTECTED]:5060gt;;tag=3D5a1c4f36BR=0Agt; gt;= Via: SIP/2.0/UDPBR=0Agt; gt; 192.168.199.121:5060;branch=3Dz9hG4bK-c87= 542-436999556-1--c87542-;rportBR=0Agt; gt; Call-ID: c90fa011e82acf3eBR= =0Agt; gt; CSeq: 1 REGISTERBR=0Agt; gt; Contact: lt;sip:[EMAIL PROTECTED] 8.199.121:5060gt;BR=0Agt; gt; Expires: 3600BR=0Agt; gt; Max-Forwar= ds: 70BR=0Agt; gt; User-Agent: FireflyBR=0Agt; gt; Content-Length: = 0BR=0Agt; gt;BR=0Agt; gt;BR=0Agt; gt; 11 headers, 0 linesBR= =0Agt; gt; Using latest request as basis requestBR=0Agt; gt; Sending = to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT):BR=0Agt; gt; = SIP/2.0 100 TryingBR=0Agt; gt; Via: SIP/2.0/UDPBR=0Agt; gt; 192.168= .199.121:5060;branch=3Dz9hG4bK-c87542-436999556-1--c87542-;rportBR=0Agt;= gt; From: lt;sip:[EMAIL PROTECTED]:5060gt;;tag=3D5a1c4f36BR=0Agt;= gt; To: lt;sip:[EMAIL PROTECTED]:5060;transport=3Dudpgt;;tag=3Das7843e= 908BR=0Agt; gt; Call-ID: c90fa011e82acf3eBR=0Agt; gt; CSeq: 1 REGIS= TERBR=0Agt; gt; User-Agent: Asterisk PBXBR=0Agt; gt; Allow: INVITE,= ACK, CANCEL, OPTIONS, BYE, REFERBR=0Agt; gt; Contact: lt;sip:[EMAIL PROTECTED] .168.199.4gt;BR=0Agt; gt; Content-Length: 0BR=0Agt; gt;BR=0Agt;= gt;BR=0Agt; gt;nbsp; to 192.168.199.121:5060BR=0Agt; gt; Transmi= tting (no NAT):BR=0Agt; gt; SIP/2.0 407 Proxy Authentication RequiredB= R=0Agt; gt; Via: SIP/2.0/UDPBR=0Agt; gt; 192.168.199.121:5060;branch= =3Dz9hG4bK-c87542-436999556-1--c87542-;rportBR=0Agt; gt; From: lt;sip:= [EMAIL PROTECTED]:5060gt;;tag=3D5a1c4f36BR=0Agt; gt; To: lt;sip:20= [EMAIL PROTECTED]:5060;transport=3Dudpgt;;tag=3Das7843e908BR=0Agt; gt; = Call-ID: c90fa011e82acf3eBR=0Agt; gt; CSeq: 1 REGISTERBR=0Agt; gt; = User-Agent: Asterisk PBXBR=0Agt; gt; Allow: INVITE, ACK, CANCEL, OPTION= S, BYE, REFERBR=0Agt; gt; Contact: lt;sip:[EMAIL PROTECTED]gt;BR= =0Agt; gt; Proxy-Authenticate: Digest realm=3Dquot;asteriskquot;, nonce= =3Dquot;38165263quot;BR=0Agt; gt; Content-Length: 0BR=0Agt; gt;B= R=0Agt; gt;BR=0Agt; gt;nbsp; to 192.168.199.121:5060BR=0Agt; gt= ; SAMPLANET1*CLIgt;BR=0Agt; gt;BR=0Agt; gt; Sip read:BR=0Agt; = gt; REGISTER sip:192.168.199.3:5060;transport=3Dudp SIP/2.0BR=0Agt; gt;= To: lt;sip:[EMAIL PROTECTED]:5060;transport=3Dudpgt;BR=0Agt; gt; Fr= om: lt;sip:[EMAIL PROTECTED]:5060gt;;tag=3D6c3de14aBR=0Agt; gt; Vi= a: SIP/2.0/UDPBR=0Agt; gt; 192.168.199.121:5060;branch=3Dz9hG4bK-c87542= -373911025-1--c87542-;rportBR=0Agt; gt; Call-ID: c90fa011e82acf3eBR= =0Agt; gt; CSeq: 1 REGISTERBR=0Agt; gt; Contact: lt;sip:[EMAIL PROTECTED] .199.121:5060gt;BR=0Agt; gt; Expires: 3600BR=0Agt; gt; Max-Forward= s: 70BR=0Agt; gt; Proxy-Authorization: DigestBR=0Agt; gt; username= =3D2003,realm=3Dquot;asteriskquot;,nonce=3Dquot;38165263quot;,uri=3Dqu= ot;sip:192.1BR=0Agt; gt; 68.199.3:5060;BR=0Agt; gt; transport=3Dudp= quot;,response=3Dquot;ec0afc0a2b13a725aa40b5c311c396d8quot;,algBR=0Ag= t; gt; orithm=3DMD5BR=0Agt; gt; User-Agent: FireflyBR=0Agt; gt; Co= ntent-Length: 0BR=0Agt; gt;BR=0Agt; gt;BR=0Agt; gt; 12 headers,= 0 linesBR=0Agt; gt; Using latest request as basis requestBR=0Agt; = gt; Sending to 192.168.199.121
Re: [Asterisk-Users] New Firefly version
Adam, works now :-) Just one further question. In my understanding Firefly's RTP Port is the SIP listen port. So there is no chance to influence the RTP/RTCP Portrange for the audio channel. Please correct me if I'm wrong. jo Adam Hart wrote: I just put up another version - fixed that issue and also added to ability to disable registration to a network. Why it's needed? If you will only be making outgoing calls but still need Firefly to use the login info for calling for lazy ppl: http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Quick run down on various ways of calling - 123 - Firefly will find the network marked as internal and dial 123 on that network +123 - Firefly will find the network marked as external and dial 123 (note no plus) on that network. [EMAIL PROTECTED] - Firefly to find the network named blah and dial 123 sip/[EMAIL PROTECTED] (Firefly will try and find the network for this one as well, otherwise make the call as 'guest') (sip:// also works) Otherwise you can use full asterisk urls eg iax/user:[EMAIL PROTECTED]/extension sip/user:[EMAIL PROTECTED]/extension jo wrote: 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Firefly's RTP port option is for RTP, not SIP listen port. All RTP goes to the one port. There's currently no option to set the SIP port (coming shortly) jo wrote: Adam, works now :-) Just one further question. In my understanding Firefly's RTP Port is the SIP listen port. So there is no chance to influence the RTP/RTCP Portrange for the audio channel. Please correct me if I'm wrong. jo Adam Hart wrote: I just put up another version - fixed that issue and also added to ability to disable registration to a network. Why it's needed? If you will only be making outgoing calls but still need Firefly to use the login info for calling for lazy ppl: http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Quick run down on various ways of calling - 123 - Firefly will find the network marked as internal and dial 123 on that network +123 - Firefly will find the network marked as external and dial 123 (note no plus) on that network. [EMAIL PROTECTED] - Firefly to find the network named blah and dial 123 sip/[EMAIL PROTECTED] (Firefly will try and find the network for this one as well, otherwise make the call as 'guest') (sip:// also works) Otherwise you can use full asterisk urls eg iax/user:[EMAIL PROTECTED]/extension sip/user:[EMAIL PROTECTED]/extension jo wrote: 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Firefly version
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) is not a local number, so that another 0 should be added (or a 9). Regards Felix -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart Sent: Monday, May 31, 2004 3:01 AM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]: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:[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 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:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Firefly version
I'm having this problem too. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 7:53 AM To: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]: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:[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 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:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
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? [EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]: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:[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 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:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] New Firefly version
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 [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 7:53 AM To: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]: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:[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 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:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Firefly version
Lol, remove the 'r' from the url. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart Sent: Monday, 31 May 2004 3:33 PM To: [EMAIL PROTECTED] 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 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... Oops, using a default stun port of 1 - fixing now :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
adam - can the g729.dll be downloaded somewhere - is this still required for g.729 support? Regards, Steven Thomas jo [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 31/05/2004 09:19 PM Please respond to asterisk-users To [EMAIL PROTECTED] cc Subject Re: [Asterisk-Users] New Firefly version 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
get http://www.virbiage.com/firefly/download/g729.zip and follow the instructions (you'll need to compile it) Steven Thomas wrote: adam - can the g729.dll be downloaded somewhere - is this still required for g.729 support? Regards, Steven Thomas *jo [EMAIL PROTECTED]* Sent by: [EMAIL PROTECTED] 31/05/2004 09:19 PM Please respond to asterisk-users To [EMAIL PROTECTED] cc Subject Re: [Asterisk-Users] New Firefly version 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
I'll look at it tomorrow jo wrote: 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
I just put up another version - fixed that issue and also added to ability to disable registration to a network. Why it's needed? If you will only be making outgoing calls but still need Firefly to use the login info for calling for lazy ppl: http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Quick run down on various ways of calling - 123 - Firefly will find the network marked as internal and dial 123 on that network +123 - Firefly will find the network marked as external and dial 123 (note no plus) on that network. [EMAIL PROTECTED] - Firefly to find the network named blah and dial 123 sip/[EMAIL PROTECTED] (Firefly will try and find the network for this one as well, otherwise make the call as 'guest') (sip:// also works) Otherwise you can use full asterisk urls eg iax/user:[EMAIL PROTECTED]/extension sip/user:[EMAIL PROTECTED]/extension jo wrote: 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
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 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... Oops, using a default stun port of 1 - fixing now :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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... Oops, using a default stun port of 1 - fixing now :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
