I was using asterisk 1.6.0.6 and the current version of asterisk gui. I manually added some extensions and now asterisk gui seems to work fine. It also is not crashing.
Thanks Eric -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of [email protected] Sent: Friday, February 27, 2009 12:00 PM To: [email protected] Subject: asterisk-gui Digest, Vol 28, Issue 26 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. asterisk 1.0.6 + asterisk gui 2.0 r4527 (Eric J. Swanson) 2. Re: asterisk 1.0.6 + asterisk gui 2.0 r4527 (Josh Thomas) 3. Re: asterisk 1.0.6 + asterisk gui 2.0 r4527 (Ryan Brindley) 4. Incoming Calling Rules - Trunk Sip (Giovanni Giusti) 5. Feature Request: Incoming Call Display BlackList (Greg Nutt) 6. Re: Feature Request: Incoming Call Display BlackList (Chuck) 7. Re: Feature Request: Incoming Call Display BlackList (Greg Nutt) 8. Re: Feature Request: Incoming Call Display BlackList (Noah Miller) ---------------------------------------------------------------------- Message: 1 Date: Thu, 26 Feb 2009 12:25:44 -0600 From: "Eric J. Swanson" <[email protected]> Subject: [asterisk-gui] asterisk 1.0.6 + asterisk gui 2.0 r4527 To: "[email protected]" <[email protected]> Message-ID: <d76cef69999d6345b81e934011b0829160a6fc4...@cenetstptrsex01.cenetcorp.lan> Content-Type: text/plain; charset="iso-8859-1" It allows me to login and then goes into an infinite loop and ends up with updating Extensions.conf and after a few minutes it goes to loading. After this loop, it comes back to the login screen The next time I attempt to log in, it says Could not connect to Server Retry. Any thoughts? Thanks Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090226/aa1d79b1/attachment.html ------------------------------ Message: 2 Date: Thu, 26 Feb 2009 14:02:10 -0500 From: Josh Thomas <[email protected]> Subject: Re: [asterisk-gui] asterisk 1.0.6 + asterisk gui 2.0 r4527 To: Asterisk GUI project discussion <[email protected]> Message-ID: <c5cc3542.6b90%[email protected]> Content-Type: text/plain; charset="iso-8859-1" Eric, Your running Asterisk 1.0.6 or something along 1.6.0? I think the 2.0 branch of the GUI works on 1.4 and 1.6 but to quote Ryan: "There is a known issue in 1.6.0.5 with action_originate in the manager interface that breaks the GUI. There is already a fix in the svn version of 1.6.0 and will be in the next minor release. You can either: use 1.6.1.x or checkout the latest svn of 1.6.0. Both should work." On 2/26/09 11:25 AM, "Eric J. Swanson" <[email protected]> wrote: It allows me to login and then goes into an infinite loop and ends up with updating Extensions.conf and after a few minutes it goes to loading. After this loop, it comes back to the login screen The next time I attempt to log in, it says Could not connect to Server Retry. Any thoughts? Thanks Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090226/036e69c4/attachment-0001.htm ------------------------------ Message: 3 Date: Thu, 26 Feb 2009 15:04:25 -0600 (CST) From: Ryan Brindley <[email protected]> Subject: Re: [asterisk-gui] asterisk 1.0.6 + asterisk gui 2.0 r4527 To: Asterisk GUI project discussion <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset="utf-8" Eric, Hrm. That cannot connect to server error is because Asterisk crashed, which wasn't a typical symptom reported by testers of the other bug Josh referred to. Although kudos to Josh for referencing that. Also, just wanted to make sure that it is in fact an infinite loop. the GUI does loop on first load because it has to submit a few config changes (to preferences.conf, http.conf, extensions.conf and dahdi_guiread.conf). It could be that something is not allowing the GUI to update those and therefore causes the loop when it reloads and checks again. If you can do the following (in order) for me and report back any changes: (1) Open up a terminal and get into the Asterisk CLI (asterisk -r). Set verbosity up (core set verbose 10). Now run the GUI like you have been and see what the last commands before the GUI crashes. (2) Verify that you have proper permissions in manager.conf. (3) edit /var/lib/asterisk/static-http/config/index.html and find the DEBUG_MODE global JS variable and set it to true (instead of false). Use Firefox and get the Firebug plugin and check for any JS errors . (4) check out the latest svn 1.6.0 and re-install asterisk from that directory (svn = subversion, of which you will need to install if it isn't already on the system): svn co http://svn.digium.com/svn/asterisk/branches/1.6.0 asterisk-1.6.0-svn (and then the normal ./configure && make && make install) I'm having you do the #4 because thats what I test with and I don't seem to have your problem :-). Plus, if the fix is already in the latest code rev, then there is no reason to re-debug an already solved issue. Hope this helps! -- 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: "Josh Thomas" <[email protected]> To: "Asterisk GUI project discussion" <[email protected]> Sent: Thursday, February 26, 2009 1:02:10 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-gui] asterisk 1.0.6 + asterisk gui 2.0 r4527 Re: [asterisk-gui] asterisk 1.0.6 + asterisk gui 2.0 r4527 Eric, Your running Asterisk 1.0.6 or something along 1.6.0? I think the 2.0 branch of the GUI works on 1.4 and 1.6 but to quote Ryan: ? There is a known issue in 1.6.0.5 with action_originate in the manager interface that breaks the GUI. There is already a fix in the svn version of 1.6.0 and will be in the next minor release. You can either: use 1.6.1.x or checkout the latest svn of 1.6.0. Both should work.? On 2/26/09 11:25 AM, "Eric J. Swanson" < [email protected] > wrote: It allows me to login and then goes into an infinite loop and ends up with updating Extensions.conf and after a few minutes it goes to loading. After this loop, it comes back to the login screen The next time I attempt to log in, it says Could not connect to Server Retry. Any thoughts? Thanks Eric _______________________________________________ --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/20090226/b8ecf81f/attachment-0001.htm ------------------------------ Message: 4 Date: Fri, 27 Feb 2009 15:16:52 +0000 From: Giovanni Giusti <[email protected]> Subject: [asterisk-gui] Incoming Calling Rules - Trunk Sip To: <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset="iso-8859-1" Hi, i have a little problem with GUI and Sip Trunk. I have set up my asterisk server 1.6.0.6 with GUI 2.0 When i call outbound OK, but when i call with my cellphone (347*******) the number of sip trunk(0574******), i recevie e fast hangup and in Debug console i see this log file. <-------------> <--- SIP read from UDP://83.211.227.21:5060 ---> INVITE sip:[email protected] SIP/2.0 Record-Route: <sip:83.211.227.21;ftag=2713899C-671;lr=on> Record-Route: <sip:83.211.227.13;ftag=2713899C-671;lr=on> Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK02bd.ae53af21.0 Via: SIP/2.0/UDP 83.211.2.218:5060;rport=56083;x-route-tag="tgrp:Slot6";branch=z9hG4bK3B6A910B From: <sip:347****[email protected]>;tag=2713899C-671 To: <sip:0574***[email protected]> Call-ID: [email protected] CSeq: 102 INVITE Max-Forwards: 8 Remote-Party-ID: <sip:347****[email protected]>;party=calling;screen=yes;privacy=off Contact: <sip:347****[email protected]:5060> Expires: 180 Content-Type: application/sdp Content-Length: 415 v=0 o=CiscoSystemsSIP-GW-UserAgent 9289 1505 IN IP4 83.211.2.218 s=SIP Call c=IN IP4 62.94.199.36 t=0 0 m=audio 63772 RTP/AVP 18 8 0 4 3 125 101 c=IN IP4 62.94.199.36 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=5.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:125 X-CCD/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (16 headers 17 lines) --- == Using SIP RTP CoS mark 5 Sending to 83.211.227.21 : 5060 (no NAT) Using INVITE request as basis request - [email protected] No user '347*******' in SIP users list Found peer 'trunk_1' for '347*******' from 83.211.227.21:5060 <--- Reliably Transmitting (no NAT) to 83.211.227.21:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 83.211.227.21;branch=0;received=83.211.227.21 Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK02bd.ae53af21.0 Via: SIP/2.0/UDP 83.211.2.218:5060;rport=56083;x-route-tag="tgrp:Slot6";branch=z9hG4bK3B6A910B From: <sip:347****[email protected]>;tag=2713899C-671 To: <sip:0574***[email protected]>;tag=as415e84f8 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4df0ff3d" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE) <--- SIP read from UDP://83.211.227.21:5060 ---> ACK sip:[email protected] SIP/2.0 Max-Forwards: 15 Record-Route: <sip:83.211.227.21;ftag=2713899C-671;lr=on> Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK02bd.ae53af21.0 From: <sip:347****[email protected]>;tag=2713899C-671 Call-ID: [email protected] To: <sip:0574***[email protected]>;tag=as415e84f8 CSeq: 102 ACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP://83.211.227.21:5060 ---> INVITE sip:[email protected] SIP/2.0 Record-Route: <sip:83.211.227.21;ftag=23BDC0D0-26A9;lr=on> Record-Route: <sip:83.211.227.13;ftag=23BDC0D0-26A9;lr=on> Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bKe114.c9f5b0a6.0 Via: SIP/2.0/UDP 62.94.71.96:5060;rport=52353;x-route-tag="tgrp:Slot7";branch=z9hG4bK45FF31981 From: <sip:347****[email protected]>;tag=23BDC0D0-26A9 To: <sip:0574***[email protected]> Call-ID: [email protected] CSeq: 102 INVITE Max-Forwards: 8 Remote-Party-ID: <sip:347****[email protected]>;party=calling;screen=yes;privacy=off Contact: <sip:347****[email protected]:5060> Expires: 180 Content-Type: application/sdp Content-Length: 434 v=0 o=CiscoSystemsSIP-GW-UserAgent 6954 399 IN IP4 62.94.71.96 s=SIP Call c=IN IP4 62.94.199.37 t=0 0 m=audio 62482 RTP/AVP 18 8 0 4 3 125 101 c=IN IP4 62.94.199.37 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=5.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:125 X-CCD/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive <-------------> --- (16 headers 18 lines) --- == Using SIP RTP CoS mark 5 Sending to 83.211.227.21 : 5060 (no NAT) Using INVITE request as basis request - [email protected] No user '347*******' in SIP users list Found peer 'trunk_1' for '347*******' from 83.211.227.21:5060 <--- Reliably Transmitting (no NAT) to 83.211.227.21:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 83.211.227.21;branch=0;received=83.211.227.21 Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bKe114.c9f5b0a6.0 Via: SIP/2.0/UDP 62.94.71.96:5060;rport=52353;x-route-tag="tgrp:Slot7";branch=z9hG4bK45FF31981 From: <sip:347****[email protected]>;tag=23BDC0D0-26A9 To: <sip:0574***[email protected]>;tag=as1d4d87a1 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4b39d7fa" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE) <--- SIP read from UDP://83.211.227.21:5060 ---> ACK sip:[email protected] SIP/2.0 Max-Forwards: 15 Record-Route: <sip:83.211.227.21;ftag=23BDC0D0-26A9;lr=on> Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bKe114.c9f5b0a6.0 From: <sip:347****[email protected]>;tag=23BDC0D0-26A9 Call-ID: [email protected] To: <sip:0574***[email protected]>;tag=as1d4d87a1 CSeq: 102 ACK Content-Length: 0 <-------------> This is a part of my configuration... users.conf [trunk_1] context=DID_trunk_1 host=voip.eutelia.it username=0574****** insecure=no secret=******** trunkname=eutelia hasiax=no registeriax=no hassip=yes registersip=yes trunkstyle=voip hasexten=no disallow=all allow=all extensions.conf [default] [DLPN_DialPlan1] include = default include = ringgroups [DID_trunk_1] include = DID_trunk_1_timeinterval_all,${timeinterval_all} include = DID_trunk_1_default [DID_trunk_1_default] [DID_trunk_1_timeinterval_all] exten = _X.,1,Goto(default,6000,1) Giovanni Giusti. _________________________________________________________________ Quali sono le pi? cliccate della settimana? http://livesearch.it.msn.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090227/859b6892/attachment-0001.htm ------------------------------ Message: 5 Date: Fri, 27 Feb 2009 11:09:17 -0500 (EST) From: Greg Nutt <[email protected]> Subject: [asterisk-gui] Feature Request: Incoming Call Display BlackList To: [email protected] Message-ID: <[email protected]> Content-Type: text/plain; charset="us-ascii" I'd like to see a feature in the gui that would allow you to create a list of DID's that one does NOT wish to proceed any further into the PBX routing. This could be applied on a per trunk or global level. Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090227/17438065/attachment-0001.htm ------------------------------ Message: 6 Date: Fri, 27 Feb 2009 08:35:05 -0800 From: "Chuck" <[email protected]> Subject: Re: [asterisk-gui] Feature Request: Incoming Call Display BlackList To: "'Asterisk GUI project discussion'" <[email protected]> Message-ID: <006901c998f9$58a978f0$09fc6a...@com> Content-Type: text/plain; charset="us-ascii" That is my next task to add DID hunt group. How are you doing this now? Chuck Coleman President CCI Technologies/CC Call Center/CSI Technologies Director of Managed Services for Gurus2go Cell 510-439-6501 CSIretro3 Confidential Email: This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. If you are not the intended recipient you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited. Please also note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of the CSI Technologies, Inc.. From: [email protected] [mailto:[email protected]] On Behalf Of Greg Nutt Sent: Friday, February 27, 2009 08:09 To: [email protected] Subject: [asterisk-gui] Feature Request: Incoming Call Display BlackList I'd like to see a feature in the gui that would allow you to create a list of DID's that one does NOT wish to proceed any further into the PBX routing. This could be applied on a per trunk or global level. Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-gui/attachments/20090227/c537770e/attachment-0001.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 3266 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-gui/attachments/20090227/c537770e/attachment-0001.jpeg ------------------------------ Message: 7 Date: Fri, 27 Feb 2009 11:48:04 -0500 (EST) From: Greg Nutt <[email protected]> Subject: Re: [asterisk-gui] Feature Request: Incoming Call Display BlackList To: Asterisk GUI project discussion <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset=us-ascii Right now I'm doing it in a sad, adhoc sort of way that isn't really worth mentioning. :p I'm rebuilding my main server however and if/when I come up with a better way of doing it I'll advise. Greg ----- Original Message ----- From: Chuck <[email protected]> Sent: Fri, 2/27/2009 11:35am To: 'Asterisk GUI project discussion' <[email protected]> Subject: Re: [asterisk-gui] Feature Request: Incoming Call Display BlackList That is my next task to add DID hunt group. How are you doing this now? Chuck Coleman President CCI Technologies/CC Call Center/CSI Technologies Director of Managed Services for Gurus2go Cell 510-439-6501 Confidential Email: This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. If you are not the intended recipient you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited. Please also note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of the CSI Technologies, Inc.. From: [email protected] [mailto:[email protected]] On Behalf Of Greg Nutt Sent: Friday, February 27, 2009 08:09 To: [email protected] Subject: [asterisk-gui] Feature Request: Incoming Call Display BlackList I'd like to see a feature in the gui that would allow you to create a list of DID's that one does NOT wish to proceed any further into the PBX routing. This could be applied on a per trunk or global level. Greg _______________________________________________ --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 ------------------------------ Message: 8 Date: Fri, 27 Feb 2009 12:44:10 -0500 From: Noah Miller <[email protected]> Subject: Re: [asterisk-gui] Feature Request: Incoming Call Display BlackList To: Asterisk GUI project discussion <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset=ISO-8859-1 Hi Greg - > I'd like to see a feature in the gui that would allow you to create a list > of DID's that one does NOT wish to proceed any further into the PBX > routing.? This could be applied on a per trunk or global level. You don't need a special feature for this, you can just create an incoming call rule to match whatever DID extensions you want, and then send it to the "hangup" destination. For example, if your normal DID range is _12XX, and you don't want 1234 to go anywhere, go to "Incoming Calling Rules", create a new incoming rule. For the pattern, put in 1234, and for the destination, select Hangup. - Noah ------------------------------ _______________________________________________ --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 28, Issue 26 ******************************************** _______________________________________________ --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
