[asterisk-gui] asterisk-gui Digest, Vol 28, Issue 26
Eric J. Swanson
eric.swanson at inetstpeters.net
Sun Mar 1 15:37:53 CST 2009
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: asterisk-gui-bounces at lists.digium.com [mailto:asterisk-gui-bounces at lists.digium.com] On Behalf Of asterisk-gui-request at lists.digium.com
Sent: Friday, February 27, 2009 12:00 PM
To: asterisk-gui at lists.digium.com
Subject: asterisk-gui Digest, Vol 28, Issue 26
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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" <eric.swanson at inetstpeters.net>
Subject: [asterisk-gui] asterisk 1.0.6 + asterisk gui 2.0 r4527
To: "asterisk-gui at lists.digium.com" <asterisk-gui at lists.digium.com>
Message-ID:
<D76CEF69999D6345B81E934011B0829160A6FC4351 at 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
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Message: 2
Date: Thu, 26 Feb 2009 14:02:10 -0500
From: Josh Thomas <josh at kickbackpoints.com>
Subject: Re: [asterisk-gui] asterisk 1.0.6 + asterisk gui 2.0 r4527
To: Asterisk GUI project discussion <asterisk-gui at lists.digium.com>
Message-ID: <C5CC3542.6B90%josh at kickbackpoints.com>
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" <eric.swanson at inetstpeters.net> 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
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Message: 3
Date: Thu, 26 Feb 2009 15:04:25 -0600 (CST)
From: Ryan Brindley <rbrindley at digium.com>
Subject: Re: [asterisk-gui] asterisk 1.0.6 + asterisk gui 2.0 r4527
To: Asterisk GUI project discussion <asterisk-gui at lists.digium.com>
Message-ID:
<7764945.184241235682264977.JavaMail.root at jupiler.digium.com>
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" <josh at kickbackpoints.com>
To: "Asterisk GUI project discussion" <asterisk-gui at lists.digium.com>
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" < eric.swanson at inetstpeters.net > 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
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Message: 4
Date: Fri, 27 Feb 2009 15:16:52 +0000
From: Giovanni Giusti <d2_won at hotmail.it>
Subject: [asterisk-gui] Incoming Calling Rules - Trunk Sip
To: <asterisk-gui at lists.digium.com>
Message-ID: <COL112-W74199CF2CC54DBB21FF746CCAA0 at phx.gbl>
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:s at 150.217.13.132 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*******@83.211.2.218>;tag=2713899C-671
To: <sip:0574******@voip.eutelia.it>
Call-ID: 4036BFDE-40A11DE-BA12BD79-214756FC at 83.211.2.218
CSeq: 102 INVITE
Max-Forwards: 8
Remote-Party-ID: <sip:347*******@83.211.2.218>;party=calling;screen=yes;privacy=off
Contact: <sip:347*******@83.211.2.218:5060>
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 - 4036BFDE-40A11DE-BA12BD79-214756FC at 83.211.2.218
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*******@83.211.2.218>;tag=2713899C-671
To: <sip:0574******@voip.eutelia.it>;tag=as415e84f8
Call-ID: 4036BFDE-40A11DE-BA12BD79-214756FC at 83.211.2.218
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 '4036BFDE-40A11DE-BA12BD79-214756FC at 83.211.2.218' in 32000 ms (Method: INVITE)
<--- SIP read from UDP://83.211.227.21:5060 --->
ACK sip:s at 150.217.13.132 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*******@83.211.2.218>;tag=2713899C-671
Call-ID: 4036BFDE-40A11DE-BA12BD79-214756FC at 83.211.2.218
To: <sip:0574******@voip.eutelia.it>;tag=as415e84f8
CSeq: 102 ACK
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP://83.211.227.21:5060 --->
INVITE sip:s at 150.217.13.132 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*******@62.94.71.96>;tag=23BDC0D0-26A9
To: <sip:0574******@voip.eutelia.it>
Call-ID: 4074D407-40A11DE-84A49CF3-13CC2153 at 62.94.71.96
CSeq: 102 INVITE
Max-Forwards: 8
Remote-Party-ID: <sip:347*******@62.94.71.96>;party=calling;screen=yes;privacy=off
Contact: <sip:347*******@62.94.71.96:5060>
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 - 4074D407-40A11DE-84A49CF3-13CC2153 at 62.94.71.96
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*******@62.94.71.96>;tag=23BDC0D0-26A9
To: <sip:0574******@voip.eutelia.it>;tag=as1d4d87a1
Call-ID: 4074D407-40A11DE-84A49CF3-13CC2153 at 62.94.71.96
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 '4074D407-40A11DE-84A49CF3-13CC2153 at 62.94.71.96' in 32000 ms (Method: INVITE)
<--- SIP read from UDP://83.211.227.21:5060 --->
ACK sip:s at 150.217.13.132 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*******@62.94.71.96>;tag=23BDC0D0-26A9
Call-ID: 4074D407-40A11DE-84A49CF3-13CC2153 at 62.94.71.96
To: <sip:0574******@voip.eutelia.it>;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/
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Message: 5
Date: Fri, 27 Feb 2009 11:09:17 -0500 (EST)
From: Greg Nutt <gregory at nutt.ca>
Subject: [asterisk-gui] Feature Request: Incoming Call Display
BlackList
To: asterisk-gui at lists.digium.com
Message-ID: <31080535.61235750957692.JavaMail.root at torrent.nutt.ca>
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
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Message: 6
Date: Fri, 27 Feb 2009 08:35:05 -0800
From: "Chuck" <pres at 2cci.com>
Subject: Re: [asterisk-gui] Feature Request: Incoming Call Display
BlackList
To: "'Asterisk GUI project discussion'"
<asterisk-gui at lists.digium.com>
Message-ID: <006901c998f9$58a978f0$09fc6ad0$@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: asterisk-gui-bounces at lists.digium.com
[mailto:asterisk-gui-bounces at lists.digium.com] On Behalf Of Greg Nutt
Sent: Friday, February 27, 2009 08:09
To: asterisk-gui at lists.digium.com
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
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Message: 7
Date: Fri, 27 Feb 2009 11:48:04 -0500 (EST)
From: Greg Nutt <gregory at nutt.ca>
Subject: Re: [asterisk-gui] Feature Request: Incoming Call Display
BlackList
To: Asterisk GUI project discussion <asterisk-gui at lists.digium.com>
Message-ID: <19462716.91235753284387.JavaMail.root at torrent.nutt.ca>
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 <pres at 2cci.com>
Sent: Fri, 2/27/2009 11:35am
To: 'Asterisk GUI project discussion' <asterisk-gui at lists.digium.com>
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: asterisk-gui-bounces at lists.digium.com
[mailto:asterisk-gui-bounces at lists.digium.com] On Behalf Of Greg Nutt
Sent: Friday, February 27, 2009 08:09
To: asterisk-gui at lists.digium.com
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
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------------------------------
Message: 8
Date: Fri, 27 Feb 2009 12:44:10 -0500
From: Noah Miller <noahisaacmiller at gmail.com>
Subject: Re: [asterisk-gui] Feature Request: Incoming Call Display
BlackList
To: Asterisk GUI project discussion <asterisk-gui at lists.digium.com>
Message-ID:
<8699dcab0902270944r3b88128cia9d0f80a11efc383 at mail.gmail.com>
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
------------------------------
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