Hi, I am using the linksys firewall. If the pix is working, then the linksys should work.
Thanks Eric ________________________________________ From: [email protected] [[email protected]] On Behalf Of [email protected] [[email protected]] Sent: Wednesday, March 04, 2009 12:00 PM To: [email protected] Subject: asterisk-gui Digest, Vol 29, Issue 5 Send asterisk-gui mailing list submissions to [email protected] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-gui or, via email, send a message with subject or body 'help' to [email protected] You can reach the person managing the list at [email protected] When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-gui digest..." Today's Topics: 1. Re: Cannot park calls (Allan Harte) 2. asterisk-gui question (Eric J. Swanson) 3. Re: asterisk-gui question (Jordan Kirby) 4. Re: asterisk-gui Call Queues (Rajesh Kumar ( Alshaya Group International )) 5. Outgoing Calling Rules (Jordan Kirby) 6. Re: Outgoing Calling Rules (Ryan Brindley) ---------------------------------------------------------------------- Message: 1 Date: Wed, 4 Mar 2009 09:03:12 -0000 From: "Allan Harte" <[email protected]> Subject: Re: [asterisk-gui] Cannot park calls To: "'Asterisk GUI project discussion'" <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset="us-ascii" OK This is the CLI log for dialling external number 123, putting on HOLD and then going to exten 700 to park. (Hope it means something to you, as it doesn't to me -- Executing [...@dlpn_dialplan1:1] [1;36;40mMacro[0;37;40m("[1;35;40mSIP/211-006e37e0[0;37;40m", "[1;35;40mtrunkdial-failover-0.3|SIP/trunk_1/123||trunk_1|[0;37;40m") in new stack -- Executing [[email protected]:1] [1;36;40mSet[0;37;40m("[1;35;40mSIP/211-006e37e0[0;37;40m", "[1;35;40mCALLERID(num)=[0;37;40m") in new stack -- Executing [[email protected]:2] [1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/211-006e37e0[0;37;40m", "[1;35;40m0?1-dial|1[0;37;40m") in new stack -- Executing [[email protected]:3] [1;36;40mSet[0;37;40m("[1;35;40mSIP/211-006e37e0[0;37;40m", "[1;35;40mCALLERID(all)=[0;37;40m") in new stack -- Executing [[email protected]:4] [1;36;40mGoto[0;37;40m("[1;35;40mSIP/211-006e37e0[0;37;40m", "[1;35;40m1-dial|1[0;37;40m") in new stack -- Goto (macro-trunkdial-failover-0.3,1-dial,1) -- Executing [[email protected]:1] [1;36;40mDial[0;37;40m("[1;35;40mSIP/211-006e37e0[0;37;40m", "[1;35;40mSIP/trunk_1/123[0;37;40m") in new stack -- Called trunk_1/123 -- SIP/trunk_1-0076f1b0 is ringing -- SIP/trunk_1-0076f1b0 answered SIP/211-006e37e0 -- Started music on hold, class 'default', on SIP/trunk_1-0076f1b0 -- Executing [...@dlpn_dialplan1:1] [1;36;40mMacro[0;37;40m("[1;35;40mSIP/211-006c7180[0;37;40m", "[1;35;40mpage|SIP/700[0;37;40m") in new stack -- Executing [...@macro-page:1] [1;36;40mChanIsAvail[0;37;40m("[1;35;40mSIP/211-006c7180[0;37;40m", "[1;35;40mSIP/700|js[0;37;40m") in new stack == Spawn extension (macro-page, s, 1) exited non-zero on 'SIP/211-006c7180' -- Stopped music on hold on SIP/trunk_1-0076f1b0 == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/211-006e37e0' in macro 'trunkdial-failover-0.3' == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/211-006e37e0' _____ From: [email protected] [mailto:[email protected]] On Behalf Of Ryan Brindley Sent: 03 March 2009 13:26 To: Asterisk GUI project discussion Subject: Re: [asterisk-gui] Cannot park calls Allan, K. Next step is to monitor the call and verify that the DIALOPTIONS are getting used. Place a call that you know isn't working and look in the Asterisk CLI (with 'core set verbose 5') for the Dial(...) line and verify that it has the tThHkK settings. -- Ryan Brindley Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA main: +1 256-428-6000 fax: +1 256-864-0464 Check us out at: http://digium.com & http://asterisk.org ----- Original Message ----- From: "Allan Harte" <[email protected]> To: "Asterisk GUI project discussion" <[email protected]> Sent: Tuesday, March 3, 2009 3:56:12 AM GMT -06:00 US/Canada Central Subject: Re: [asterisk-gui] Cannot park calls Hi The features page has all the Dial Options selected. Extensions.conf has DIALOPTIONS = tThHkK set in the [Globals] context -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090304/f53d937c/attachment-0001.htm ------------------------------ Message: 2 Date: Wed, 4 Mar 2009 07:13:14 -0600 From: "Eric J. Swanson" <[email protected]> Subject: [asterisk-gui] asterisk-gui question To: "[email protected]" <[email protected]> Message-ID: <d76cef69999d6345b81e934011b0829160a6fc4...@cenetstptrsex01.cenetcorp.lan> Content-Type: text/plain; charset="iso-8859-1" Hi, I am seeing time out when connecting to a asterisk gui website. Has anyone tried using the gui with the computer behind a firewall? These time outs are occuring when I attempt to connect to the gui from a computer outside of the firewall. I do not have port 80 or 443 natted to the asterisk computer. The only port I have natted are the sip ports, rtp and 8088, which is part of the asterisk-gui url. Thanks Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090304/b2e8654a/attachment-0001.htm ------------------------------ Message: 3 Date: Wed, 4 Mar 2009 13:34:50 +0000 From: Jordan Kirby <[email protected]> Subject: Re: [asterisk-gui] asterisk-gui question To: Asterisk GUI project discussion <[email protected]> Message-ID: <68edef55066b0c44bceb2ebcc863ead3c7c27...@pr-gn-exch-01> Content-Type: text/plain; charset="us-ascii" We use the web interface behind our Cisco PIX firewall all the time, both it nat and non-nat environments. We just have an ACL in place to allow TCP/8088 traffic through. What firewall are you using? Jordan From: [email protected] [mailto:[email protected]] On Behalf Of Eric J. Swanson Sent: 04 March 2009 13:13 To: [email protected] Subject: [asterisk-gui] asterisk-gui question Hi, I am seeing time out when connecting to a asterisk gui website. Has anyone tried using the gui with the computer behind a firewall? These time outs are occuring when I attempt to connect to the gui from a computer outside of the firewall. I do not have port 80 or 443 natted to the asterisk computer. The only port I have natted are the sip ports, rtp and 8088, which is part of the asterisk-gui url. Thanks Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090304/5aacc95f/attachment-0001.htm ------------------------------ Message: 4 Date: Wed, 4 Mar 2009 17:48:19 +0300 From: "Rajesh Kumar ( Alshaya Group International )" <[email protected]> Subject: Re: [asterisk-gui] asterisk-gui Call Queues To: 'Asterisk GUI project discussion' <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset="us-ascii" Hey Gurus, One more thing I think GUI lacks is periodic announcement on Call Queues for example announcement to caller that they are first in the queue you will be connected to next available agent, your call is important for us etc. etc. any thoughts? Regards, Rajesh Kumar [email protected]<mailto:[email protected]> Alshaya Group International Tel: +965 298-0555 Ext.- 307 Mobile: +965 722-4083 From: [email protected] [mailto:[email protected]] On Behalf Of Jordan Kirby Sent: Wednesday, March 04, 2009 4:35 PM To: Asterisk GUI project discussion Subject: Re: [asterisk-gui] asterisk-gui question We use the web interface behind our Cisco PIX firewall all the time, both it nat and non-nat environments. We just have an ACL in place to allow TCP/8088 traffic through. What firewall are you using? Jordan From: [email protected] [mailto:[email protected]] On Behalf Of Eric J. Swanson Sent: 04 March 2009 13:13 To: [email protected] Subject: [asterisk-gui] asterisk-gui question Hi, I am seeing time out when connecting to a asterisk gui website. Has anyone tried using the gui with the computer behind a firewall? These time outs are occuring when I attempt to connect to the gui from a computer outside of the firewall. I do not have port 80 or 443 natted to the asterisk computer. The only port I have natted are the sip ports, rtp and 8088, which is part of the asterisk-gui url. Thanks Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090304/0a0c2173/attachment-0001.htm ------------------------------ Message: 5 Date: Wed, 4 Mar 2009 15:02:45 +0000 From: Jordan Kirby <[email protected]> Subject: [asterisk-gui] Outgoing Calling Rules To: Asterisk GUI project discussion <[email protected]> Message-ID: <68edef55066b0c44bceb2ebcc863ead3c7c27...@pr-gn-exch-01> Content-Type: text/plain; charset="us-ascii" Hi, I'm running Asterisk SVN-branch-1.6.1-r179256 and GUI r4546. When I try to go to "Outgoing Calling Rules" my browser just sits waiting for the javascript (a view source shows the html has all loaded). I can't find anything in firebug other than it gets to the end of astman.js but doesn't seem to start on callingrules.js, although I think that may be firebug not reporting it correctly as if I remove the reference to callingrules.js from callingrules.html the page loads (obviously without any functionality). I get the same behaviour in IE, Firefox and Chrome. As firebug doesn't seem to see callingrules.js I can't see how to put a breakpoint in (any ideas?) Has anyone else encountered this? Thanks Jordan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090304/d00cf0fa/attachment-0001.htm ------------------------------ Message: 6 Date: Wed, 4 Mar 2009 09:18:34 -0600 (CST) From: Ryan Brindley <[email protected]> Subject: Re: [asterisk-gui] Outgoing Calling Rules To: Asterisk GUI project discussion <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset="utf-8" Jordan, I haven't ran across this yet, but two things you might try doing: turning on debug mode in index.html and, using 1.6.0 instead of 1.6.1, unless of course there is some specific feature only in 1.6.1. The times that I've seen the GUI hang has been because of some unreported GUI error. Unfortunately the GUI uses try blocks, but doesn't always report the errors :-/. Debug mode usually reports most of the errors. -- Ryan Brindley Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA main: +1 256-428-6000 fax: +1 256-864-0464 Check us out at: http://digium.com & http://asterisk.org ----- Original Message ----- From: "Jordan Kirby" <[email protected]> To: "Asterisk GUI project discussion" <[email protected]> Sent: Wednesday, March 4, 2009 9:02:45 AM GMT -06:00 US/Canada Central Subject: [asterisk-gui] Outgoing Calling Rules Hi, I'm running Asterisk SVN-branch-1.6.1-r179256 and GUI r4546. When I try to go to "Outgoing Calling Rules" my browser just sits waiting for the javascript (a view source shows the html has all loaded). I can't find anything in firebug other than it gets to the end of astman.js but doesn't seem to start on callingrules.js, although I think that may be firebug not reporting it correctly as if I remove the reference to callingrules.js from callingrules.html the page loads (obviously without any functionality). I get the same behaviour in IE, Firefox and Chrome. As firebug doesn't seem to see callingrules.js I can't see how to put a breakpoint in (any ideas?) Has anyone else encountered this? Thanks Jordan _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090304/88138195/attachment-0001.htm ------------------------------ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui End of asterisk-gui Digest, Vol 29, Issue 5 ******************************************* _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui
