Re: [asterisk-users] Hints realtime table structure Ast 11
On 2016-05-18 16:32, Neeraj Chand wrote: Hi All, Has anyone used hints in realtime ? (As in storing and loading hints from odbc) I cannot find a table structure for this anywhere...? Thanks Neeraj Hints are defined in the dialplan so if you are loading your dialplan from a database it is the same thing. I personally use static realtime for my dialplan as I find the "real" realtime scheme very inefficient because you still have to modify the dialplan text file to insert switches (which for me defeats the purpose). -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez dCAP #1349 +52 (55)9116-91161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints realtime table structure Ast 11
Hi All, Has anyone used hints in realtime ? (As in storing and loading hints from odbc) I cannot find a table structure for this anywhere...? Thanks Neeraj -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hints and server-side DND (do not disturb)
--- On Wed, 4/18/12, Warren Selby wcse...@selbytech.com wrote: exten = *280,n,Set(DEVICE_STATE(Custom:lamp)=BUSY) Thanks! So in short, it's all about DEVICE_STATE or DEVSTATE for * 1.4. I've just one last issue and was wondering how to run the following command on a remote Asterisk server: Set(DEVSTATE(Custom:mycustomstate)=BUSY) ie. how can I set a DEVICE STATE from one Asterisk server to another (for clustering purposes). Can I do it via AMI by running something like this? Setvar(DEVSTATE(Custom:mycustomstate)=BUSY) Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hints and server-side DND (do not disturb)
Hi, Currently I'm using hints to determine SIP presence. As I understand it, a SIP extension can be labeled as busy, ringing, etc, based on a channel status. So a channel MUST be present. If it isn't then the extension is considered to be available. If my statement is correct then is there a way to set the extesnion as busy even if there's no channel associated with this extension? eg. when an extension sets server-side DND (Do Not Disturb), it actually sets a boolean value in astdb. Whenever asterisk tries to route a call to this extension, it first checks this value. Obviously, there's no way I can use hints in this scenario, or is there? Is it possible to somehow create a dummy channel whenever an extension sets server-side DND (custom context) and delete it whenever it unsets it? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hints and server-side DND (do not disturb)
יעע -Original Message- From: Vieri rentor...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 17 Apr 2012 23:27:10 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] hints and server-side DND (do not disturb) Hi, Currently I'm using hints to determine SIP presence. As I understand it, a SIP extension can be labeled as busy, ringing, etc, based on a channel status. So a channel MUST be present. If it isn't then the extension is considered to be available. If my statement is correct then is there a way to set the extesnion as busy even if there's no channel associated with this extension? eg. when an extension sets server-side DND (Do Not Disturb), it actually sets a boolean value in astdb. Whenever asterisk tries to route a call to this extension, it first checks this value. Obviously, there's no way I can use hints in this scenario, or is there? Is it possible to somehow create a dummy channel whenever an extension sets server-side DND (custom context) and delete it whenever it unsets it? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hints and server-side DND (do not disturb)
On Wed, Apr 18, 2012 at 1:27 AM, Vieri rentor...@yahoo.com wrote:
Hi,
Currently I'm using hints to determine SIP presence. As I understand it, a
SIP extension can be labeled as busy, ringing, etc, based on a channel
status. So a channel MUST be present. If it isn't then the extension is
considered to be available.
If my statement is correct then is there a way to set the extesnion as
busy even if there's no channel associated with this extension?
eg. when an extension sets server-side DND (Do Not Disturb), it actually
sets a boolean value in astdb. Whenever asterisk tries to route a call to
this extension, it first checks this value. Obviously, there's no way I can
use hints in this scenario, or is there? Is it possible to somehow create a
dummy channel whenever an extension sets server-side DND (custom
context) and delete it whenever it unsets it?
I've done something similar using night-mode type logic. All calls
coming into the system first do a check against the db to see if night-mode
is enabled or not. If it is, route calls to voicemail, if it's not, route
calls normally. You can also use custom hints to set busy lamps on
appropriate phones. The receptionist can then hit the monitored button on
her phone to turn on or turn off night-mode. Here's some snippets from
existing dialplan...
[mainmenu]
; Main IVR
exten = s,1,Verbose(Inbound call to main number - checking if night mode
or normal)
exten = s,n,Set(NIGHTMODE=${DB(nightmode/enable)})
exten = s,n,GotoIf($[${NIGHTMODE} = 1]?nightmode)
exten = s,n,Verbose(Normal mode - Proceeding Normally)
exten = s,n,...
exten = s,n,...
exten = s,n,...
exten = s,n(nightmode),Verbose(Night mode - going straight to voicemail)
exten = s,n,Voicemail(@default,su)
exten = s,n,Hangup()
[internal]
; Night Mode
exten = *280,1,Answer()
exten = *280,n,GotoIf($[${DB(nightmode/enable)} = 1]?disable:enable)
exten = *280,n(enable),Verbose(Enabling night mode)
exten = *280,n,Set(DB(nightmode/enable)=1)
exten = *280,n,Set(DEVICE_STATE(Custom:lamp)=BUSY)
exten = *280,n,Playback(enabled)
exten = *280,n,Hangup()
exten = *280,n(disable),Verbose(Disabling night mode)
exten = *280,n,Set(DB(nightmode/enable)=0)
exten = *280,n,Set(DEVICE_STATE(Custom:lamp)=NOT_INUSE)
exten = *280,n,Playback(disabled)
exten = *280,n,Hangup()
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints problem - NAT problem?
Hi all, I try to figure out why I have empty : sip show subscriptions list in may asterisk 1.6. When device is registering to asterisk I can see in log: NOTICE[25603]: chan_sip.c:21518 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1010 but sip show subscriptions is just empty. May it be the problem because devices are registering to asterisk from behind NAT? I belive this is the cause why hints does not work in my dialplan. Regards, Jarek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints custom:abcdef
Hello list, I want certain devices to monitor certain extensions/SIPaccounts and other devices to monitor other extensions/SIPaccounts. Therefore I do the following : [from-TEST1] include = test1-blf [from-TEST2] include = test2-blf [test1-blf] exten = 10,hint,SIP/testcorp1 exten = 20,hint,SIP/testcorp2 [test2-blf] exten = 10,hint,SIP/testcorp110 exten = 20,hint,SIP/testcorp120 SIPaccounts with a context definition of from-TEST1 can not monitor the extensions of test2-blf. SIPaccounts with a context definition of from-TEST2 can not monitor the extensions of test1-blf. This works great. But now I have a problem with custom hints. When I do the following : [test1-blf] exten = 10,hint,SIP/testcorp1 exten = 20,hint,SIP/testcorp2 exten = 80,hint,custom:light1 I see that SIPaccounts which enter the dialplan in context [from-TEST2] also see the state (Green/Red) of hints defined in test1-blf. So how can I make a difference between custom hints in one context and custom hints in another context ?? Is there something like : Set(DEVSTATE(*Custom:light1@test1-blf*)=INUSE)* * ?? Or can there be only one custom:light1 label ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints custom:abcdef
In other words : is it correct to say that hints need to be unique, even if they are defined in different contexts ? On 05/20/2011 12:07 PM, Jonas Kellens wrote: Hello list, I want certain devices to monitor certain extensions/SIPaccounts and other devices to monitor other extensions/SIPaccounts. Therefore I do the following : [from-TEST1] include = test1-blf [from-TEST2] include = test2-blf [test1-blf] exten = 10,hint,SIP/testcorp1 exten = 20,hint,SIP/testcorp2 [test2-blf] exten = 10,hint,SIP/testcorp110 exten = 20,hint,SIP/testcorp120 SIPaccounts with a context definition of from-TEST1 can not monitor the extensions of test2-blf. SIPaccounts with a context definition of from-TEST2 can not monitor the extensions of test1-blf. This works great. But now I have a problem with custom hints. When I do the following : [test1-blf] exten = 10,hint,SIP/testcorp1 exten = 20,hint,SIP/testcorp2 exten = 80,hint,custom:light1 I see that SIPaccounts which enter the dialplan in context [from-TEST2] also see the state (Green/Red) of hints defined in test1-blf. So how can I make a difference between custom hints in one context and custom hints in another context ?? Is there something like : Set(DEVSTATE(*Custom:light1@test1-blf*)=INUSE)* * ?? Or can there be only one custom:light1 label ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hints through a Local channel
I am actually deploying on a 1.6.1.6 but it does not seem to work - maybe I
am using a wrong syntax?.
pbx-ch*CLI core show version
Asterisk 1.6.1.6 built by root @ pbx-ch on a i686 running Linux on
2009-09-11 16:54:55 UTC
I see this works:
exten = 100,hint,SIP/${EXTENSION}
pbx-ch*CLI core show hint 100
1...@agents : SIP/100 State:Unavailable Watchers 0
but this does not
exten = 100,hint,SIP/${DB(/Agent/100)}
pbx-ch*CLI core show hint 100
1...@agents : SIP/ State:Unavailable Watchers 0
Thanks
l.
l.
2009/12/14 Tilghman Lesher tles...@digium.com
On Monday 14 December 2009 03:20:11 pm Stephen Davies wrote:
On 12/14/09, Lenz Emilitri lenz.lo...@gmail.com wrote:
But more dynamical, so I would try and look up the actual channel in
the
AstDB, like:
exten = XXX,hint,${DB(myagent/${EXTEN})}
This does not seem to be working - is there a way to work around this
You can't use functions in a hint exten.
That's no longer true as of 1.6.1. If the OP is using 1.6.0, however, the
solution becomes clear.
--
Tilghman
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Loway - home of QueueMetrics - http://queuemetrics.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hints through a Local channel
Thanks that's exactly what I was looking for! I had seen a patch for it but did not notice this was in the main trunk. l. 2009/12/14 Stephen Davies stephen.l.dav...@gmail.com What you are missing is the new state-interface parameter to AddQueueMember. You can't use functions in a hint exten. Steve -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hints through a Local channel
Hello all,
I am trying to set up a dynamic channel to be used as an Agent dialer for a
queue - you know, trying to replace AgentCallBackLogin for an Asterisk 1.6.
I would like to do something like:
[myagents]
exten = XXX,1,Set(realchan=${DB(myagent/${EXTEN})})
exten = XXX,n,Dial(${realchan},tT,60)
This basically fetches the actual channel to be used for dialling and dials
it. What I would like now is to make app_queue aware in advance of the state
of each channel, something like:
exten = 100,hint,SIP/705 (and this works)
But more dynamical, so I would try and look up the actual channel in the
AstDB, like:
exten = XXX,hint,${DB(myagent/${EXTEN})}
This does not seem to be working - is there a way to work around this issue?
(I admit this is the fist time I'm trying to use devoce state and the
related functions, so maybe there is a very simple slution right in front of
my big nose and I'm not seeing it).
Thanks a lot for your help,
l.
--
Loway - home of QueueMetrics - http://queuemetrics.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hints through a Local channel
What you are missing is the new state-interface parameter to AddQueueMember.
You can't use functions in a hint exten.
Steve
On 12/14/09, Lenz Emilitri lenz.lo...@gmail.com wrote:
Hello all,
I am trying to set up a dynamic channel to be used as an Agent dialer for a
queue - you know, trying to replace AgentCallBackLogin for an Asterisk 1.6.
I would like to do something like:
[myagents]
exten = XXX,1,Set(realchan=${DB(myagent/${EXTEN})})
exten = XXX,n,Dial(${realchan},tT,60)
This basically fetches the actual channel to be used for dialling and dials
it. What I would like now is to make app_queue aware in advance of the state
of each channel, something like:
exten = 100,hint,SIP/705 (and this works)
But more dynamical, so I would try and look up the actual channel in the
AstDB, like:
exten = XXX,hint,${DB(myagent/${EXTEN})}
This does not seem to be working - is there a way to work around this issue?
(I admit this is the fist time I'm trying to use devoce state and the
related functions, so maybe there is a very simple slution right in front of
my big nose and I'm not seeing it).
Thanks a lot for your help,
l.
--
Loway - home of QueueMetrics - http://queuemetrics.com
--
Sent from my mobile device
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hints through a Local channel
On Monday 14 December 2009 03:20:11 pm Stephen Davies wrote:
On 12/14/09, Lenz Emilitri lenz.lo...@gmail.com wrote:
But more dynamical, so I would try and look up the actual channel in the
AstDB, like:
exten = XXX,hint,${DB(myagent/${EXTEN})}
This does not seem to be working - is there a way to work around this
You can't use functions in a hint exten.
That's no longer true as of 1.6.1. If the OP is using 1.6.0, however, the
solution becomes clear.
--
Tilghman
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hints (was: Re: Manager ExtensionState function)
Azher Mughal schrieb: Now when a call is connected i can see Idle shouldn't be 'In Use' : *CLI show hints -= Registered Asterisk Dial Plan Hints =- 3...@demo: SIP/8172 State:IdleWatchers 0 - 1 hints registered I have qualify=yes for all the SIP providers in sip.conf. Is there some other setting that needs to be done for this to work properly. http://translate.google.com/translate?js=nprev=_thl=deie=UTF-8u=http%3A%2F%2Fwww.das-asterisk-buch.de%2F2.1%2Fblf-leds.htmlsl=detl=enhistory_state0= Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hints
Thanks. Philipp Kempgen wrote: Azher Mughal schrieb: Now when a call is connected i can see Idle shouldn't be 'In Use' : *CLI show hints -= Registered Asterisk Dial Plan Hints =- 3...@demo: SIP/8172 State:IdleWatchers 0 - 1 hints registered I have qualify=yes for all the SIP providers in sip.conf. Is there some other setting that needs to be done for this to work properly. http://translate.google.com/translate?js=nprev=_thl=deie=UTF-8u=http%3A%2F%2Fwww.das-asterisk-buch.de%2F2.1%2Fblf-leds.htmlsl=detl=enhistory_state0= Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints
Running an earlier version of Asterisk (1.2), we were using Hints to show busy extensions on other (SNOM) phones. When we went to version 1.4 they stopped working, using the same syntax. (Copied and pasted) Does anyone have any tips or clues? Is the exact location in the file critical? Maybe we put the code in a back alley? Cary ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints
The hints may have been moved to a different context. On Polycom phones, the hint either has to be in the default context or specified in the directory (1...@somecontext). _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Monday, March 09, 2009 2:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints Running an earlier version of Asterisk (1.2), we were using Hints to show busy extensions on other (SNOM) phones. When we went to version 1.4 they stopped working, using the same syntax. (Copied and pasted) Does anyone have any tips or clues? Is the exact location in the file critical? Maybe we put the code in a back alley? Cary ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints
To get busy state for a sip channel in 1.4 it appears the peer/friend must have a call-limit. Steve On 3/9/09, Cary Fitch ca...@usawide.net wrote: Running an earlier version of Asterisk (1.2), we were using Hints to show busy extensions on other (SNOM) phones. When we went to version 1.4 they stopped working, using the same syntax. (Copied and pasted) Does anyone have any tips or clues? Is the exact location in the file critical? Maybe we put the code in a back alley? Cary -- Sent from my mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints
According to voip-info.org, the call-limit is mandatory to make hints work as of 1.4.X. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Davies Sent: Monday, March 09, 2009 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints To get busy state for a sip channel in 1.4 it appears the peer/friend must have a call-limit. Steve On 3/9/09, Cary Fitch ca...@usawide.net wrote: Running an earlier version of Asterisk (1.2), we were using Hints to show busy extensions on other (SNOM) phones. When we went to version 1.4 they stopped working, using the same syntax. (Copied and pasted) Does anyone have any tips or clues? Is the exact location in the file critical? Maybe we put the code in a back alley? Cary -- Sent from my mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints
Thanks to all for the hints about Hints. Got them working. Shoulda Read the Fine Manual. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, March 09, 2009 4:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints According to voip-info.org, the call-limit is mandatory to make hints work as of 1.4.X. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Davies Sent: Monday, March 09, 2009 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints To get busy state for a sip channel in 1.4 it appears the peer/friend must have a call-limit. Steve On 3/9/09, Cary Fitch ca...@usawide.net wrote: Running an earlier version of Asterisk (1.2), we were using Hints to show busy extensions on other (SNOM) phones. When we went to version 1.4 they stopped working, using the same syntax. (Copied and pasted) Does anyone have any tips or clues? Is the exact location in the file critical? Maybe we put the code in a back alley? Cary -- Sent from my mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
Valid question. The problem (hints not working) was reported to me by 3 customers within the same 48 hours. I hadn`t changed anything for a while, but I do remember having removed call-limits on the SIP phonesabout 3 weeks ago. Guess nobody missed hints for a while, hence my incorrect statement about having changed nothing. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, November 27, 2008 13:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints stopped working suddently Mike, I don't want to be a smart ass, but (as you claimed) if you didn't change anything I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. how was it working before ? I really want to know, as there may be something else going on in the background. Julian. Mike wrote: Just to follow-up, because this may one day be found by someone with the same issue, I fixed this: My problem was that my sip peers did not have a call-limit setup. For some (unknown to me) reason, hints only work for peers with a call-limit defined (if using realtime, that would mean something numerical, and not NULL). Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 11:21 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Yes I did. Nothing changes, really. And it all looks good. What I don't get is why the status unavailable appears when the phone is disconnected, but the status inuse doesn't when on a call. That unavailable works fine is some sort of proof that everything is setup properly Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 11:18 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Have you tried doing core show hints and sip show peers before and after asterisk restart to see what if anything changes? *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 10:11 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 8:51 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 6:33 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, * * * * *Mike* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email
Re: [asterisk-users] Hints stopped working suddently
On 1.4.22.1 the call-limit is a required parameter for hints to work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Friday, November 28, 2008 10:03 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-CommercialDiscussion' Subject: Re: [asterisk-users] Hints stopped working suddently Valid question. The problem (hints not working) was reported to me by 3 customers within the same 48 hours. I hadn`t changed anything for a while, but I do remember having removed call-limits on the SIP phonesabout 3 weeks ago. Guess nobody missed hints for a while, hence my incorrect statement about having changed nothing. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Thursday, November 27, 2008 13:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints stopped working suddently Mike, I don't want to be a smart ass, but (as you claimed) if you didn't change anything I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. how was it working before ? I really want to know, as there may be something else going on in the background. Julian. Mike wrote: Just to follow-up, because this may one day be found by someone with the same issue, I fixed this: My problem was that my sip peers did not have a call-limit setup. For some (unknown to me) reason, hints only work for peers with a call-limit defined (if using realtime, that would mean something numerical, and not NULL). Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 11:21 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Yes I did. Nothing changes, really. And it all looks good. What I don't get is why the status unavailable appears when the phone is disconnected, but the status inuse doesn't when on a call. That unavailable works fine is some sort of proof that everything is setup properly. Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 11:18 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Have you tried doing core show hints and sip show peers before and after asterisk restart to see what if anything changes? *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 10:11 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 8:51 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 6:33 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, * * * * *Mike* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users
Re: [asterisk-users] Hints stopped working suddently
Just to follow-up, because this may one day be found by someone with the same issue, I fixed this: My problem was that my sip peers did not have a call-limit setup. For some (unknown to me) reason, hints only work for peers with a call-limit defined (if using realtime, that would mean something numerical, and not NULL). Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 11:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Yes I did. Nothing changes, really. And it all looks good. What I don't get is why the status unavailable appears when the phone is disconnected, but the status inuse doesn't when on a call. That unavailable works fine is some sort of proof that everything is setup properly Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 11:18 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Have you tried doing core show hints and sip show peers before and after asterisk restart to see what if anything changes? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 10:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 8:51 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 6:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
Mike, I don't want to be a smart ass, but (as you claimed) if you didn't change anything I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. how was it working before ? I really want to know, as there may be something else going on in the background. Julian. Mike wrote: Just to follow-up, because this may one day be found by someone with the same issue, I fixed this: My problem was that my sip peers did not have a call-limit setup. For some (unknown to me) reason, hints only work for peers with a call-limit defined (if using realtime, that would mean something numerical, and not NULL). Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 11:21 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Yes I did. Nothing changes, really. And it all looks good. What I don't get is why the status unavailable appears when the phone is disconnected, but the status inuse doesn't when on a call. That unavailable works fine is some sort of proof that everything is setup properly… Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 11:18 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Have you tried doing “core show hints” and “sip show peers” before and after asterisk restart to see what if anything changes? *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 10:11 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 8:51 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 6:33 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, * * * * *Mike* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 6:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints stopped working suddently
Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
For me, the Polycom loses its subscription when asterisk is restarted. However, as long as the phone is restarted after asterisk, everything works fine. Worth a look. (I'm running a Polycom 500, so my firmware is older than yours.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
The phone should renew itself to asterisk periodically even after a reboot. My setup renews the connection every 2 minutes (non-critical, small shop). _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Wednesday, November 26, 2008 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints stopped working suddently For me, the Polycom loses its subscription when asterisk is restarted. However, as long as the phone is restarted after asterisk, everything works fine. Worth a look. (I'm running a Polycom 500, so my firmware is older than yours.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
Not at all, I do everything with vi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 8:51 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 6:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
Good theory, but I had already tried that (and my phone re-subscribes every 60 seconds anyways) so that's not it. Regards, Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Wednesday, November 26, 2008 10:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints stopped working suddently For me, the Polycom loses its subscription when asterisk is restarted. However, as long as the phone is restarted after asterisk, everything works fine. Worth a look. (I'm running a Polycom 500, so my firmware is older than yours.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
Have you tried doing core show hints and sip show peers before and after asterisk restart to see what if anything changes? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 10:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 8:51 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 6:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
Yes I did. Nothing changes, really. And it all looks good. What I don't get is why the status unavailable appears when the phone is disconnected, but the status inuse doesn't when on a call. That unavailable works fine is some sort of proof that everything is setup properly Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 11:18 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Have you tried doing core show hints and sip show peers before and after asterisk restart to see what if anything changes? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 10:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 8:51 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 6:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints and realtime
On Monday 17 November 2008 10:38:30 am Julian Lyndon-Smith wrote:
Is it possible to use hints from a realtime source like a db or curl ?
I was looking at the grandstream GXP2000 Expansion Module (EXT) which
has 56 fully programmable keys that work with BLF. You can daisy-chain 2
of these together to get 112 keys, plus the 18 on the 2010 phone to give
130 potential blfs.
What I was wanting to do was to use the BLF as a type of agent monitor
as well - when the agent logs in, they go green (from blank), and when
they log out for them to go blank. I would assume that I can do this by
adding / removing the hints from extensions.conf, and was wondering if
it were possible to configure the hints from a realtime source.
Failing that, I could always dialplan reload from the login / logout
extensions, but that seems a little messy.
You cannot use hints from a database, no. However, as of 1.6.0, you can do
something almost as good: you can use a pattern match in extensions.conf
and use a function such as ${CURL()} to map the extension and context to a
particular device.
--
Tilghman
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints and realtime
Is it possible to use hints from a realtime source like a db or curl ? I was looking at the grandstream GXP2000 Expansion Module (EXT) which has 56 fully programmable keys that work with BLF. You can daisy-chain 2 of these together to get 112 keys, plus the 18 on the 2010 phone to give 130 potential blfs. What I was wanting to do was to use the BLF as a type of agent monitor as well - when the agent logs in, they go green (from blank), and when they log out for them to go blank. I would assume that I can do this by adding / removing the hints from extensions.conf, and was wondering if it were possible to configure the hints from a realtime source. Failing that, I could always dialplan reload from the login / logout extensions, but that seems a little messy. Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints / State change on outgoing calls
On 9/19/07, Alex Epshteyn [EMAIL PROTECTED] wrote: Also, Asterisk restart results in all the watchers being lost. Is there a way to force the phone to subscribe to notifications after restart (short of rebooting it) and is it phone specific? Usually resubscribe-interval for extensions is client controlled, much like SIP re-register interval. Just make sure it's in between the min and max registration times as displayed in the output of 'sip show settings', otherwise you can run into problems where the phone thinks that the subscription is valid for longer than Asterisk does. -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints / State change on outgoing calls
Hi, I am trying to set BLF on SNOM phones. With call-limit=4 in sip.conf and hints in the extensions.conf a call to the extension correctly shows state as InUse (show hints) and BLF works. When call is originated from the extension the associated state remains Idle, so no notification and no BLF. Is there something else that has to be set for state to change (and watchers notified) on the outgoing calls? Also, Asterisk restart results in all the watchers being lost. Is there a way to force the phone to subscribe to notifications after restart (short of rebooting it) and is it phone specific? This is Asterisk 1.4.11. Thanks, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints and Noop
Hello, I want to get rid of bunch of useless notices in the logs when the hint is not found, does setting the hint to noop for everything breaks anything? exten = _X.,hint,NoOp So far it did what I wanted. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints
I used to run Asterisk 1.4.4 but had to revert back to 1.2.13 to minimize a bug we were coming across. 1.4.5 looked promising, but the hints are broken and making it so I'll likely have to go back to 1.2.13 until I get the hints fixed. I'm using Grandstream phones hints on the parked extensions. I should also clarify that when I upgraded versions, I renamed all Asterisk folders (/var/log /var/lib /usr/lib /var/spool) so I could have a 'clean' install of 1.4.5. There's a few things that are happening on hints. First, on a fresh reboot, despite the server saying all hints are IDLE the Grandstream phones light up as if INUSE. This has never happened across umpteen different versions of Asterisk I've ran. The fix is to actually put the parked extensions INUSE and clear them, then they function fine...for a bit. The second problem is, after about an hour, hints just stop working. Well, hints actually work, but the phones stop watching. [EMAIL PROTECTED] : park:[EMAIL PROTECTED] State:IdleWatchers 29 [EMAIL PROTECTED] : park:[EMAIL PROTECTED] State:IdleWatchers 29 [EMAIL PROTECTED] : park:[EMAIL PROTECTED] State:IdleWatchers 29 [EMAIL PROTECTED] : park:[EMAIL PROTECTED] State:IdleWatchers 29 [EMAIL PROTECTED] : park:[EMAIL PROTECTED] State:IdleWatchers 29 The watchers in an hour or so after a fresh reboot will drop to 0, I believe it has to do with when the phone reregisters. Which brings me to the third problem (directly related to a phone reregistering). After a fresh reboot, if I reboot all phones before any calls get parked, all phones work properly (for an hour anyway). However, if I reboot a phone *after* a calls been placed on hold, the hints do not work for that phone and the Watchers doesn't get updated (say I have Watches:28 and I plug another phone in, it should go to 29 but it won't unless I restart Asterisk). So somewhere I've got something messed up. Not sure where to look, it seems odd that as soon as the parking lot is used (and a hint updated) it kills any new watchers from attaching, as well as all watchers drop off after an hour. Any thoughts on where to look? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hints howto reset if wrong subscription Asterisk 1.2
Hi, sometimes Asterisk told me in the subscription: status confirmed so LED is on if the softphone is disconnected or the registration has expired. So the whole weekend LEDs have the wrong status. Manager Command Extensionstate is working correct, only the subscription is wrong. How can I reset this by hand? SIP clients are in relatime, dialplan is txt file. I tried to delete astdb and fields in the mySQL database, but without success. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hints not working using SVN-branch-1.4-r59289
Just wanted to update the group. I copied the config file to a working server and the hints worked without any problems. Can anyone tell me if they have a working system using hits and SVN-branch-1.4-r59289 or newer. Eric Hall From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Tuesday, April 03, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hints not working using SVN-branch-1.4-r59289 Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI show hints -= Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 2 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 4 --- - 6 hints registered Here is the sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes; Notify subscriptions on HOLD state (default: no) limitonpeers=yes allow=ulaw [21] ;Bill Salmons type=peer username=21 callerid=Bill Salmons 21 secret=21 host=dynamic context=default mailbox=21 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=30 [23] ;Teresa Trautman type=peer username=23 callerid=Teresa Trautman 23 secret=23 host=dynamic context=default mailbox=23 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [25] ;Bill Goldsmith type=peer username=25 callerid=Bill Goldsmith 25 secret=25 host=dynamic context=default mailbox=25 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [26] ;Joelle Harris type=peer username=26 callerid=Joelle Harris 26 secret=26 host=dynamic context=default mailbox=26 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [29] ;Amanda Anderson type=peer username=29 callerid=Amanda Anderson 29 secret=29 host=dynamic context=default mailbox=29 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [30] ;Joelle Harris type=peer username=30 callerid=Liz Williamson 30 secret=30 host=dynamic context=default mailbox=30 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [ata] type=peer username=ata host=dynamic context=default secret=ata here is the extensions.conf [default] include = parkedcalls exten = 21,hint(SIP/21) exten = 21,1,answer exten = 21,n,dial(sip/21|30|kw) exten = 21,n,voicemail([EMAIL PROTECTED]|u) exten = 23,hint(sip/23) exten = 23,1,answer exten = 23,n,dial(sip/23|30|kw) exten = 23,n,voicemail([EMAIL PROTECTED]|u) exten = 25,hint(SIP/25) exten = 25,1,answer exten = 25,n,dial(sip/25|30|kw) exten = 25,n,voicemail([EMAIL PROTECTED]|u) exten = 26,hint(SIP/26) exten = 26,1,answer exten = 26,n,dial(sip/26|30|kw) exten = 26,n,voicemail([EMAIL PROTECTED]|u) exten = 29,hint(SIP/29) exten = 29,1,answer exten = 29,n,dial(sip/29|30|kw) exten = 29,n,voicemail([EMAIL PROTECTED]|u) exten = 30,hint(SIP/30) exten = 30,1,answer exten = 30,n,dial(sip/30|30|kw) exten = 30,n,voicemail([EMAIL PROTECTED]|u) -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.25/744 - Release Date: 4/3/2007 5:32 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Hints not working using SVN-branch-1.4-r59289 **** FIXED ****
Just wanted to update the list I found the problem. In my extensions.conf I had exten = 21,hint(SIP/21) It should be exten = 21,hint,SIP/21 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Wednesday, April 04, 2007 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Hints not working using SVN-branch-1.4-r59289 Just wanted to update the group. I copied the config file to a working server and the hints worked without any problems. Can anyone tell me if they have a working system using hits and SVN-branch-1.4-r59289 or newer. Eric Hall From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Tuesday, April 03, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hints not working using SVN-branch-1.4-r59289 Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI show hints -= Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 2 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 4 --- - 6 hints registered Here is the sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes; Notify subscriptions on HOLD state (default: no) limitonpeers=yes allow=ulaw [21] ;Bill Salmons type=peer username=21 callerid=Bill Salmons 21 secret=21 host=dynamic context=default mailbox=21 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=30 [23] ;Teresa Trautman type=peer username=23 callerid=Teresa Trautman 23 secret=23 host=dynamic context=default mailbox=23 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [25] ;Bill Goldsmith type=peer username=25 callerid=Bill Goldsmith 25 secret=25 host=dynamic context=default mailbox=25 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [26] ;Joelle Harris type=peer username=26 callerid=Joelle Harris 26 secret=26 host=dynamic context=default mailbox=26 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [29] ;Amanda Anderson type=peer username=29 callerid=Amanda Anderson 29 secret=29 host=dynamic context=default mailbox=29 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [30] ;Joelle Harris type=peer username=30 callerid=Liz Williamson 30 secret=30 host=dynamic context=default mailbox=30 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [ata] type=peer username=ata host=dynamic context=default secret=ata here is the extensions.conf [default] include = parkedcalls exten = 21,hint(SIP/21) exten = 21,1,answer exten = 21,n,dial(sip/21|30|kw) exten = 21,n,voicemail([EMAIL PROTECTED]|u) exten = 23,hint(sip/23) exten = 23,1,answer exten = 23,n,dial(sip/23|30|kw) exten = 23,n,voicemail([EMAIL PROTECTED]|u) exten = 25,hint(SIP/25) exten = 25,1,answer exten = 25,n,dial(sip/25|30|kw) exten = 25,n,voicemail([EMAIL PROTECTED]|u) exten = 26,hint(SIP/26) exten = 26,1,answer exten = 26,n,dial(sip/26|30|kw) exten = 26,n,voicemail([EMAIL PROTECTED]|u) exten = 29,hint(SIP/29) exten = 29,1,answer exten = 29,n,dial(sip/29|30|kw) exten = 29,n,voicemail([EMAIL PROTECTED]|u) exten = 30,hint(SIP/30) exten = 30,1,answer exten = 30,n,dial(sip/30|30|kw) exten = 30,n,voicemail([EMAIL PROTECTED]|u) -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446
[asterisk-users] Hints not working using SVN-branch-1.4-r59289
Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI show hints -= Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 2 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 4 --- - 6 hints registered Here is the sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes; Notify subscriptions on HOLD state (default: no) limitonpeers=yes allow=ulaw [21] ;Bill Salmons type=peer username=21 callerid=Bill Salmons 21 secret=21 host=dynamic context=default mailbox=21 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=30 [23] ;Teresa Trautman type=peer username=23 callerid=Teresa Trautman 23 secret=23 host=dynamic context=default mailbox=23 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [25] ;Bill Goldsmith type=peer username=25 callerid=Bill Goldsmith 25 secret=25 host=dynamic context=default mailbox=25 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [26] ;Joelle Harris type=peer username=26 callerid=Joelle Harris 26 secret=26 host=dynamic context=default mailbox=26 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [29] ;Amanda Anderson type=peer username=29 callerid=Amanda Anderson 29 secret=29 host=dynamic context=default mailbox=29 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [30] ;Joelle Harris type=peer username=30 callerid=Liz Williamson 30 secret=30 host=dynamic context=default mailbox=30 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [ata] type=peer username=ata host=dynamic context=default secret=ata here is the extensions.conf [default] include = parkedcalls exten = 21,hint(SIP/21) exten = 21,1,answer exten = 21,n,dial(sip/21|30|kw) exten = 21,n,voicemail([EMAIL PROTECTED]|u) exten = 23,hint(sip/23) exten = 23,1,answer exten = 23,n,dial(sip/23|30|kw) exten = 23,n,voicemail([EMAIL PROTECTED]|u) exten = 25,hint(SIP/25) exten = 25,1,answer exten = 25,n,dial(sip/25|30|kw) exten = 25,n,voicemail([EMAIL PROTECTED]|u) exten = 26,hint(SIP/26) exten = 26,1,answer exten = 26,n,dial(sip/26|30|kw) exten = 26,n,voicemail([EMAIL PROTECTED]|u) exten = 29,hint(SIP/29) exten = 29,1,answer exten = 29,n,dial(sip/29|30|kw) exten = 29,n,voicemail([EMAIL PROTECTED]|u) exten = 30,hint(SIP/30) exten = 30,1,answer exten = 30,n,dial(sip/30|30|kw) exten = 30,n,voicemail([EMAIL PROTECTED]|u) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints on FXO channels
I am trying to get a couple phones (GXP-2000 and Aastra 9133i) to monitor an FXO port. If I do something like: 9,hint,Zap/9 I can see that when I do show hints it is listed and it does change status when it is inuse or idle. But no matter how I configure the phones I always get Watchers 0. I have tried changing the hint number in case you need more digits but still the same result. Is it possible to see the hints for FXO ports from a phone? -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chàvez Prats Director de Tecnologìa +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms
Mark was working on this, I think it was called sla and it called something line apperance On 11/21/06, John Lange [EMAIL PROTECTED] wrote: Hints are not working in 1.4b3 period. Snom 360s do not show any status updates. However, before you file a bug report you might want to check to see if there are changes to the way hints are implemented in 1.4. It might be a configuration problem rather than a bug but I have not had time to look into it. John On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote: Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip debug' shows absolutely no NOTIFY or XML presence messages as calls progress.. Next stop Mantis :-) CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms
Thanks, John - this confirms what we are seeing. 'show hints' output isn't changing, so it looks like a bug. I'll open one and see what happens. A. On Nov 21, 2006, at 5:44 PM, John Lange wrote: Hints are not working in 1.4b3 period. Snom 360s do not show any status updates. However, before you file a bug report you might want to check to see if there are changes to the way hints are implemented in 1.4. It might be a configuration problem rather than a bug but I have not had time to look into it. John On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote: Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip debug' shows absolutely no NOTIFY or XML presence messages as calls progress.. Next stop Mantis :-) CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms
http://bugs.digium.com/view.php?id=8405 On Nov 22, 2006, at 9:11 AM, Anthony Rodgers wrote: Thanks, John - this confirms what we are seeing. 'show hints' output isn't changing, so it looks like a bug. I'll open one and see what happens. A. On Nov 21, 2006, at 5:44 PM, John Lange wrote: Hints are not working in 1.4b3 period. Snom 360s do not show any status updates. However, before you file a bug report you might want to check to see if there are changes to the way hints are implemented in 1.4. It might be a configuration problem rather than a bug but I have not had time to look into it. John On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote: Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip debug' shows absolutely no NOTIFY or XML presence messages as calls progress.. Next stop Mantis :-) CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms
Hints are not working in 1.4b3 period. Snom 360s do not show any status updates. However, before you file a bug report you might want to check to see if there are changes to the way hints are implemented in 1.4. It might be a configuration problem rather than a bug but I have not had time to look into it. John On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote: Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip debug' shows absolutely no NOTIFY or XML presence messages as calls progress.. Next stop Mantis :-) CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hints causing hang in reload
I have a system right now that has 32 extensions that I am setting up hints for snip exten = 4521,hint,SIP/4521 exten = 4522,hint,SIP/4522 exten = 4523,hint,SIP/4523 exten = 4524,hint,SIP/4524 exten = 4525,hint,SIP/4525 /snip The problem that I am running into is when I issue a reload, it hangs for about 30-40 seconds before completing the reload. I have found that by taking the hints out of the config it reloads immediately as expected. Has anyone else encountered this? Is there a decent explanation? Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints to help me debug cdr_odbc not inserting
Has anyone got a hint as to how I can best debug my problem with writing cdr to an odbc database? Problem: It doesn't insert records, it doesn't complain either . The cdr entries turn up in cdr_csv/Master.csv just fine, but not in my database. debug log says: Jul 18 16:57:07 DEBUG[25511] cdr_odbc.c: cdr_odbc: Logging uniqueid Jul 18 16:57:07 DEBUG[25511] cdr_odbc.c: cdr_odbc: Logging in GMT which it didn't till I had the dsn correct. I can use isql to insert entries into the database, but asterisk doesn't seem to want to. I have the config in flat files, so I get warnings from res_odbc, but that isn't the problem (is it?) The (only) thing I'm doing that's out of the ordinary is that I'm using Oracle XE (on a remote machine) as the database and using Oracle's unixOdbc driver. I had to work on the postgres version of the table create code to get something equivalent in oracle. How can I see why asterisk isn't attempting an insert ? Or if it is, why don't I see any errors? Thanks in advance for any clues. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hints/subscriptions accross IAX
Faris Raouf wrote: But I need to get an LED to light up on a GS in Location2 when a line on the Polycom at Location1 is in use. Is this possible? If so, can anybody give me any pointers as to how? Not at this time, no. There has been talk of building a method for doing this, but so far there is nothing in Asterisk itself. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hints/subscriptions accross IAX
(I hope this isn't html - Thunderbird is so annoying) I'm new to using hints/subscriptions on * so please be patient with me. I have two * systems in different geographic locations, connected via IAX Location1 has a Polycom 600 and a GXP-2000 phone Location 2 has a single GXP-2000. With the latest GS firmware, at Location1 I've managed to get an LED to light up on the GS phone when a line on the Polycom is in use. This is great. But I need to get an LED to light up on a GS in Location2 when a line on the Polycom at Location1 is in use. Is this possible? If so, can anybody give me any pointers as to how? Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hints and busy lamps for phones registered on SER
We use SER to front several Asterisk systems. Phones register on SER, which also acts as a load balancing and failover proxy for the Asterisks. Phone account details are held in MySQL, which Asterisk could access but does not currently do so. At present, call routing is done on the Asterisks using a FastAGI program which does the database access. We've been asked to implement busy lamps. Looking at voip-info, it seems that all hint configuration is handled statically, and worse needs to have dedicated channel names for each phone which won't work with phones registered on SER. Does anyone know of any facilities for handling this scenario? -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HINTS with Polycom stops working after asterisk reload
Dear Users, Recently I have started using HINT option in Asterisk 1.2.4 with my Polycom 500 phone. What I have notice that for a day or two everything is working great but then HINTs stop working on my Polycom phone. It also happens when I reload asterisk from console. I do sip debug and I do not see anymore asterisk sending NOTIFY messages about my watched extension. To make it work again I need to restart my Polycom phone. Is it a bug or I am missing some configuration? Thank you in advance. Bartosz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HINTS with Polycom stops working after asterisk reload
Hi ter Can you help me in configuring SIP phone in Asterisk for Progressive dial Thank You Have Great Day Sathvik On Tue, 25 Apr 2006 Bartosz Jozwiak wrote : Dear Users, Recently I have started using HINT option in Asterisk 1.2.4 with my Polycom 500 phone. What I have notice that for a day or two everything is working great but then HINTs stop working on my Polycom phone. It also happens when I reload asterisk from console. I do sip debug and I do not see anymore asterisk sending NOTIFY messages about my watched extension. To make it work again I need to restart my Polycom phone. Is it a bug or I am missing some configuration? Thank you in advance. Bartosz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HINTS with Polycom stops working after aster isk reload
Me too, with 3X Snom 360's in CAP positions. I just gave up in the end and cron'd a SIP REBOOT from my * box at strategic times. If your phone supports the REBOOT directive I'd just reboot your phone every day, 4 AM or what have you. -Original Message- From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Sent: Monday, April 24, 2006 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] HINTS with Polycom stops working after asterisk reload Dear Users, Recently I have started using HINT option in Asterisk 1.2.4 with my Polycom 500 phone. What I have notice that for a day or two everything is working great but then HINTs stop working on my Polycom phone. It also happens when I reload asterisk from console. I do sip debug and I do not see anymore asterisk sending NOTIFY messages about my watched extension. To make it work again I need to restart my Polycom phone. Is it a bug or I am missing some configuration? Thank you in advance. Bartosz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HINTS with Polycom stops working after asterisk reload
I think a 'sip reload' will keep your sip subscriptions. -Original Message- From: Colin Anderson [mailto:[EMAIL PROTECTED] Sent: Monday, April 24, 2006 1:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] HINTS with Polycom stops working after asterisk reload Me too, with 3X Snom 360's in CAP positions. I just gave up in the end and cron'd a SIP REBOOT from my * box at strategic times. If your phone supports the REBOOT directive I'd just reboot your phone every day, 4 AM or what have you. -Original Message- From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Sent: Monday, April 24, 2006 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] HINTS with Polycom stops working after asterisk reload Dear Users, Recently I have started using HINT option in Asterisk 1.2.4 with my Polycom 500 phone. What I have notice that for a day or two everything is working great but then HINTs stop working on my Polycom phone. It also happens when I reload asterisk from console. I do sip debug and I do not see anymore asterisk sending NOTIFY messages about my watched extension. To make it work again I need to restart my Polycom phone. Is it a bug or I am missing some configuration? Thank you in advance. Bartosz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HINTS with Polycom stops working after asterisk reload
Douglas Garstang wrote: I think a 'sip reload' will keep your sip subscriptions. It will now, yes. The OP said he was using Asterisk 1.2.4, which was released long before this bug was fixed. That's why it usually wise to update to the latest release before posting a question like this to the list :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hints in Realtime
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Do hints work in Realtime asterisk? not finding much on the list archives or anywhere else for that matter... I have tried using -1 priority as mentioned once or twice but no joy Thought? -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFEI/M2y9wPyZpnL2URAnSjAJ9yZdWfxu7pncgbiGWCutXO8+Y55QCgqsR9 q+sgMrusZWUKdRWINK+ZeQI= =6gsq -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hints between servers?
Chris Bagnall wrote: Greetings all, Has anyone managed to get dialplan status hints working across multiple servers? I've separated a load of SIP users out across 2 servers today, but it'd be useful if they could still see each others' status. I've replaced the various hint lines for the sip devices now on another box with: exten = 210,hint,IAX2/otherserver/210 Where 210 is defined on the other server as follows: exten = 210,hint,SIP/210 All of them report state as unavailable when doing a show hints in the dialplan. Have I got the syntax wrong, or is this something that's not meant to work in the first place? yes, it would be a cool feature. At this point, IAX2 does not support any device status reports at all, as far as I know. I know that Alan implemented something in Firefly a while ago for it. We could implement this in SIP, by forcing an outbound subscription, but if all the servers are Asterisk servers there has to be more simple ways to solve this as well as cross-server voicemail notification. /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hints between servers?
We could implement this in SIP, by forcing an outbound subscription, but if all the servers are Asterisk servers there has to be more simple ways to solve this as well as cross-server voicemail notification. Could you elaborate on that please? I'm almost certain to come across the cross-server voicemail notification issue at the same time - any way around that one? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hints between servers?
Greetings all, Has anyone managed to get dialplan status hints working across multiple servers? I've separated a load of SIP users out across 2 servers today, but it'd be useful if they could still see each others' status. I've replaced the various hint lines for the sip devices now on another box with: exten = 210,hint,IAX2/otherserver/210 Where 210 is defined on the other server as follows: exten = 210,hint,SIP/210 All of them report state as unavailable when doing a show hints in the dialplan. Have I got the syntax wrong, or is this something that's not meant to work in the first place? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hints between servers?
Chris Bagnall wrote: All of them report state as unavailable when doing a show hints in the dialplan. Have I got the syntax wrong, or is this something that's not meant to work in the first place? The latter... cross-server device state is not implemented. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hints between servers?
You should be able to throw an OpenSER box in between the two Asterisk systems, and with a bit of configuration get it working. We do something similar, except the SER box sits between the phones and Asterisk. It passes the SUBSCRIBE/NOTIFY messages backwards and forwards between the two. Doug. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hints between servers? Chris Bagnall wrote: All of them report state as unavailable when doing a show hints in the dialplan. Have I got the syntax wrong, or is this something that's not meant to work in the first place? The latter... cross-server device state is not implemented. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hints on hardware to use
Hello, I have to deploy an Asterisk PBX with this requirements: - 1 or 2 ISDN lines in input/output - 14 internal analog phones (yes, I know, analog ones... ;( ) - Billing interface for the operator (for usage of analog phones) For the external interface I'm thinking about Beronet Quad Span ISDN/BRI Card TE/NT (BN4S0). I'm in trouble about the internal interfaces: the first thought was about Digium card (4 x TDM400P with 2 x S100M modules each), but it's difficult to find a MB with 5 PCI, and I'll have no chanches for future expansions. Does anyone of you know a PCI card with 8 FXS port that SURELY works with Asterisk? I'm ready to examine any other piece of hardware with 8 or more FXS ports, too... By the way, for billing operations I'm going to check AstBill sofware; did anyone positively try it with asterisk in operational environment? Any hint will be greatly appreciated... ;) Thanks Jonathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Hints on hardware to use
Hello, I have to deploy an Asterisk PBX with this requirements: - 1 or 2 ISDN lines in input/output - 14 internal analog phones (yes, I know, analog ones... ;( ) - Billing interface for the operator (for usage of analog phones) For the external interface I'm thinking about Beronet Quad Span ISDN/BRI Card TE/NT (BN4S0). I'm in trouble about the internal interfaces: the first thought was about Digium card (4 x TDM400P with 2 x S100M modules each), but it's difficult to find a MB with 5 PCI, and I'll have no chanches for future expansions. Does anyone of you know a PCI card with 8 FXS port that SURELY works with Asterisk? I'm ready to examine any other piece of hardware with 8 or more FXS ports, too... By the way, for billing operations I'm going to check AstBill sofware; did anyone positively try it with asterisk in operational environment? Any hint will be greatly appreciated... ;) Thanks Jonathan For your internal analog extensions why not get a Digium T1 card and a channel bank. I only have experience with Adtran 600E but they work extremely well and can be had used on ebay for about $600 regularly (if you are lucky you may be able to get it much cheaper.) I read the Rhino channel banks are much cheaper and work well with asterisk but have no personal experience. With the Channel bank solution, you are looking at $500 for the T1 board and another $600 for the channel bank with 24 FXS ports. Its a solid solution and gives you tons of room to upgrade from 14 FXS ports to 24 by simply adding phones. Thanks, Steve Totaro ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Hints on hardware to use
On Wednesday 19 October 2005 15:34, asterisk wrote: Hello, I have to deploy an Asterisk PBX with this requirements: - 1 or 2 ISDN lines in input/output - 14 internal analog phones (yes, I know, analog ones... ;( ) - Billing interface for the operator (for usage of analog phones) For the external interface I'm thinking about Beronet Quad Span ISDN/BRI Card TE/NT (BN4S0). I'm in trouble about the internal interfaces: the first thought was about Digium card (4 x TDM400P with 2 x S100M modules each), but it's difficult to find a MB with 5 PCI, and I'll have no chanches for future expansions. Does anyone of you know a PCI card with 8 FXS port that SURELY works with Asterisk? I'm ready to examine any other piece of hardware with 8 or more FXS ports, too... By the way, for billing operations I'm going to check AstBill sofware; did anyone positively try it with asterisk in operational environment? Jonathan For your internal analog extensions why not get a Digium T1 card and a channel bank. I only have experience with Adtran 600E but they work extremely well and can be had used on ebay for about $600 regularly (if you are lucky you may be able to get it much cheaper.) I read the Rhino channel banks are much cheaper and work well with asterisk but have no personal experience. With the Channel bank solution, you are looking at $500 for the T1 board and another $600 for the channel bank with 24 FXS ports. Its a solid solution and gives you tons of room to upgrade from 14 FXS ports to 24 by simply adding phones. Thanks, Steve Totaro It's surely a better way than mine to solve the problem... But how can I integrate Adtran 600E with Asterisk box (apart the phisical connection with the two T1 ports)? How changes the configuration of Asterisk files? Is it like a bridge across two T1 lines or what? I'm not so expert about this type of Asterisk configurations: can I find hints or docs somewhere? I've also googled a bit, and I've found the MOSA3716 box: 16 FXS and 2 ethernet ports, for about 1.200$ at bobascom. What do you think about it? It seems * compatible, and with ethernet ports it wouldn't need anything else than 1 ISDN card for inbound/outgoing calls... It could be completely transparent to * box and the analog phones... Have you ever heard something about it? Thanks in advance Jonathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Hints and Call Waiting
Hi! We have a big problem in our call center: when an agent does an outgoing call it can receive calls from the queues. The same happens if one agent transfer a call for another agent... and the ringing tone while in a call is puting the agents like crazy... We have the hints working with lines like this in extensions.conf: exten = 101,hint,SIP/101 If we set incominglimit to 1 the agent cannot do another call (to do attended transfers) We are using Beta1 Can anyone help? Thanks, Joao Antunes ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Hints and Call Waiting
We use Cisco phones and we simply disabled call-waiting for those lines. Don't know if that will help, but whatever soft/hardphone you are using probably has a way to disable call-waiting. Tom On Oct 15, 2005, at 5:38 AM, João Paulo Antunes wrote: Hi! We have a big problem in our call center: when an agent does an outgoing call it can receive calls from the queues. The same happens if one agent transfer a call for another agent... and the ringing tone while in a call is puting the agents like crazy... We have the hints working with lines like this in extensions.conf: exten = 101,hint,SIP/101 If we set incominglimit to 1 the agent cannot do another call (to do attended transfers) We are using Beta1 Can anyone help? Thanks, Joao Antunes ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] hints and the sNOM 360
Hi Paul It's working for me ! (CVS-HEAD 1.0.9 FC3) I'm using the snom 360IP with firmware 4.2 http://www.snom.com/download/snom360-4.2-SIP-j.bin In my extensions.conf I have: exten = 100,hint,SIP/100 ; SIP Phone 100 exten = 101,hint,SIP/101 ; SIP Phone 101 exten = 102,hint,SIP/102 ; SIP Phone 102 On my phone I used the same setup as You. A good hint is: Be patient. It often takes up to 5 min. before it starts working for me. Normally I start *, start snom, start other phones. Hope this is of any use ! Reg. BennyB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hewlett Sent: 19. september 2005 18:49 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] hints and the sNOM 360 Hi I am trying to get a SNOM 360 to monitor other extensions i.e. when someone makes a call to/from another extension, one of the LED's on the SNOM 360 will change state. I am using 1.0.9/bristuff-8l. I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the relevant articles on the wiki on 'hints' and also on the 'devstate' app. I set the first function key on the 360 to extension 2001 - this transforms itself into sip:[EMAIL PROTECTED];user=phone when I save it. The function key type is set to 'Destination' as recommended by a number of articles on the Wiki. aside This seems to contradict the 360 manual which states that the function key type should be set to 'Line'. /aside In the dialplan I put [myhints] exten = 2001,hint,sip/2001 exten = 2001,1,macro(stdexten,sip/2001) exten = 2001,2,hangup In sip.conf I have [2001] type=friend username=2001 subscribecontext=myhints host=dynamic mailbox=2001 callerid=ext 2001 incominglimit=1 [2002] type=friend username=2002 subscribecontext=myhints host=dynamic mailbox=2001 callerid=SNOM360 2002 I restart asterisk from scratch and then reboot the 360. The * console shows one entry when typing the command 'sip show subscriptions' which looks correct. Inspection of the sip trace log on the 360's web page reveals that the registration succeeds and that the subscription of the 2001 from the 360 also gets a 200 OK reply. However when I dial into extension 2001 nothing happens to the led's on the 360. Inspection of the 'sip trace log' on the 360's web page reveals that it does not receive any NOTIFY from asterisk. I am at my wits end - anybody got any ideas ? Paul HE ~ -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] hints and the sNOM 360
Oh yeah, And: Turn OFF Filter Packets from Registrar. Turn ON Support Broken Registrar. This may or may not be a security risk, but for testing, it will help to see if toggling these make a difference. Shanon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shanon Swafford Sent: Thursday, September 22, 2005 12:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] hints and the sNOM 360 SIP Message Reference: # Reboot Phone which is 2000 monitoring 2001s state: UA--- SUBSCRIBE ---Asterisk UA--- 200 OK ---Asterisk # Asterisk saves subscription: # Wait for a call: # Call Comes to 2001: # Asterisk should realize somehow that it needs to NOTIFY 2000 about the call. UA--- NOTIFY ---Asterisk UA--- 200 OK ---Asterisk # Phone light flashes # 2001 Answers call UA--- NOTIFY ---Asterisk UA--- 200 OK ---Asterisk # Phone LED goes steady # Then another NOTIFY to turn off the LED when the call is over. # This works on my Snom 4S SIP Proxy. I don't know much about the Asterisk side, but I do know that the monitoring phone should receive a NOTIFY when the monitored phone is called. Until Asterisk sends the NOTIFY, it won't work. If you watch the sip debug output on *, does it show * sending a NOTIFY when the call is made? Is it to the correct IP and PORT that the phone is listening on? This might NOT be 5060. If not, the new question is why:) You try to register your phone to 2001@ip_of_asterisk instead of asterisk. Maybe * acts different that way. If you do this, make sure to redo your destination so that the phone will auto fill [EMAIL PROTECTED] rather than keeping the existing [EMAIL PROTECTED] Last, see this message from Monday: http://lists.digium.com/pipermail/asterisk-users/2005-September/126034.html Is that a coincedence? Shanon ABP Technology -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hewlett Sent: Monday, September 19, 2005 11:49 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] hints and the sNOM 360 Hi I am trying to get a SNOM 360 to monitor other extensions i.e. when someone makes a call to/from another extension, one of the LED's on the SNOM 360 will change state. I am using 1.0.9/bristuff-8l. I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the relevant articles on the wiki on 'hints' and also on the 'devstate' app. I set the first function key on the 360 to extension 2001 - this transforms itself into sip:[EMAIL PROTECTED];user=phone when I save it. The function key type is set to 'Destination' as recommended by a number of articles on the Wiki. aside This seems to contradict the 360 manual which states that the function key type should be set to 'Line'. /aside In the dialplan I put [myhints] exten = 2001,hint,sip/2001 exten = 2001,1,macro(stdexten,sip/2001) exten = 2001,2,hangup In sip.conf I have [2001] type=friend username=2001 subscribecontext=myhints host=dynamic mailbox=2001 callerid=ext 2001 incominglimit=1 [2002] type=friend username=2002 subscribecontext=myhints host=dynamic mailbox=2001 callerid=SNOM360 2002 I restart asterisk from scratch and then reboot the 360. The * console shows one entry when typing the command 'sip show subscriptions' which looks correct. Inspection of the sip trace log on the 360's web page reveals that the registration succeeds and that the subscription of the 2001 from the 360 also gets a 200 OK reply. However when I dial into extension 2001 nothing happens to the led's on the 360. Inspection of the 'sip trace log' on the 360's web page reveals that it does not receive any NOTIFY from asterisk. I am at my wits end - anybody got any ideas ? Paul HE ~ -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] hints and the sNOM 360
Hi I am trying to get a SNOM 360 to monitor other extensions i.e. when someone makes a call to/from another extension, one of the LED's on the SNOM 360 will change state. I am using 1.0.9/bristuff-8l. I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the relevant articles on the wiki on 'hints' and also on the 'devstate' app. I set the first function key on the 360 to extension 2001 - this transforms itself into sip:[EMAIL PROTECTED];user=phone when I save it. The function key type is set to 'Destination' as recommended by a number of articles on the Wiki. aside This seems to contradict the 360 manual which states that the function key type should be set to 'Line'. /aside In the dialplan I put [myhints] exten = 2001,hint,sip/2001 exten = 2001,1,macro(stdexten,sip/2001) exten = 2001,2,hangup In sip.conf I have [2001] type=friend username=2001 subscribecontext=myhints host=dynamic mailbox=2001 callerid=ext 2001 incominglimit=1 [2002] type=friend username=2002 subscribecontext=myhints host=dynamic mailbox=2001 callerid=SNOM360 2002 I restart asterisk from scratch and then reboot the 360. The * console shows one entry when typing the command 'sip show subscriptions' which looks correct. Inspection of the sip trace log on the 360's web page reveals that the registration succeeds and that the subscription of the 2001 from the 360 also gets a 200 OK reply. However when I dial into extension 2001 nothing happens to the led's on the 360. Inspection of the 'sip trace log' on the 360's web page reveals that it does not receive any NOTIFY from asterisk. I am at my wits end - anybody got any ideas ? Paul HE ~ -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] hints and the sNOM 360
SIP Message Reference: # Reboot Phone which is 2000 monitoring 2001s state: UA--- SUBSCRIBE ---Asterisk UA--- 200 OK ---Asterisk # Asterisk saves subscription: # Wait for a call: # Call Comes to 2001: # Asterisk should realize somehow that it needs to NOTIFY 2000 about the call. UA--- NOTIFY ---Asterisk UA--- 200 OK ---Asterisk # Phone light flashes # 2001 Answers call UA--- NOTIFY ---Asterisk UA--- 200 OK ---Asterisk # Phone LED goes steady # Then another NOTIFY to turn off the LED when the call is over. # This works on my Snom 4S SIP Proxy. I don't know much about the Asterisk side, but I do know that the monitoring phone should receive a NOTIFY when the monitored phone is called. Until Asterisk sends the NOTIFY, it won't work. If you watch the sip debug output on *, does it show * sending a NOTIFY when the call is made? Is it to the correct IP and PORT that the phone is listening on? This might NOT be 5060. If not, the new question is why:) You try to register your phone to 2001@ip_of_asterisk instead of asterisk. Maybe * acts different that way. If you do this, make sure to redo your destination so that the phone will auto fill [EMAIL PROTECTED] rather than keeping the existing [EMAIL PROTECTED] Last, see this message from Monday: http://lists.digium.com/pipermail/asterisk-users/2005-September/126034.html Is that a coincedence? Shanon ABP Technology -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hewlett Sent: Monday, September 19, 2005 11:49 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] hints and the sNOM 360 Hi I am trying to get a SNOM 360 to monitor other extensions i.e. when someone makes a call to/from another extension, one of the LED's on the SNOM 360 will change state. I am using 1.0.9/bristuff-8l. I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the relevant articles on the wiki on 'hints' and also on the 'devstate' app. I set the first function key on the 360 to extension 2001 - this transforms itself into sip:[EMAIL PROTECTED];user=phone when I save it. The function key type is set to 'Destination' as recommended by a number of articles on the Wiki. aside This seems to contradict the 360 manual which states that the function key type should be set to 'Line'. /aside In the dialplan I put [myhints] exten = 2001,hint,sip/2001 exten = 2001,1,macro(stdexten,sip/2001) exten = 2001,2,hangup In sip.conf I have [2001] type=friend username=2001 subscribecontext=myhints host=dynamic mailbox=2001 callerid=ext 2001 incominglimit=1 [2002] type=friend username=2002 subscribecontext=myhints host=dynamic mailbox=2001 callerid=SNOM360 2002 I restart asterisk from scratch and then reboot the 360. The * console shows one entry when typing the command 'sip show subscriptions' which looks correct. Inspection of the sip trace log on the 360's web page reveals that the registration succeeds and that the subscription of the 2001 from the 360 also gets a 200 OK reply. However when I dial into extension 2001 nothing happens to the led's on the 360. Inspection of the 'sip trace log' on the 360's web page reveals that it does not receive any NOTIFY from asterisk. I am at my wits end - anybody got any ideas ? Paul HE ~ -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] hints not working on CVS HEAD
i've tried it on both snom190 and eyeBeam none of them work. nothing is changed in configs. is there any success in making snom LEDs work on CVS HEAD? thanks, paradise dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] hints and polycom IP 300 phones
Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600 Is there any additional debug apart from show hints to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =- 655 : SIP/gs102_1 State 0 Watchers 0 605 : Zap/127 State 0 Watchers 3 604 : SIP/ata186_2 State 0 Watchers 0 603 : SIP/ata186_1 State 0 Watchers 0 602 : Zap/129 State 0 Watchers 0 601 : SIP/polycom_b State 0 Watchers 1 600 : SIP/polycom_a State 1 Watchers 2 The IP600 is watching 605 and 600 and working nicely for both, the IP300 is watching 601, but isn't working Has anyone got a IP300 phone to display the status ?? Any suggestions for things to look at/etc ?? PS, of course, the current state is that 600 is off-hook and all others are on-hook. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 8304 0001www.websitemanagers.com.au ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] hints and polycom IP 300 phones
Hello, I have two polycom ip300. I patched Asterisk However it don't show status of phones when I press busy, Away, ... So I use Sip Express Router (proxy sip) for IM and Presence SIMPLE. Harry --- Adam Goryachev [EMAIL PROTECTED] a écrit : Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600 Is there any additional debug apart from show hints to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =- 655 : SIP/gs102_1 State 0 Watchers 0 605 : Zap/127 State 0 Watchers 3 604 : SIP/ata186_2 State 0 Watchers 0 603 : SIP/ata186_1 State 0 Watchers 0 602 : Zap/129 State 0 Watchers 0 601 : SIP/polycom_b State 0 Watchers 1 600 : SIP/polycom_a State 1 Watchers 2 The IP600 is watching 605 and 600 and working nicely for both, the IP300 is watching 601, but isn't working Has anyone got a IP300 phone to display the status ?? Any suggestions for things to look at/etc ?? PS, of course, the current state is that 600 is off-hook and all others are on-hook. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 8304 0001 www.websitemanagers.com.au ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Hints: What are they? How do they work?
We have lots of customers who want to be able to look at their Cisco 79XX phone and see lines are in use. Do hints work with Cisco phones? Perhaps someone can clarify this: We have many 7960's. Which means 6 lines right? Not really, cause each line must have a SIP username/password and must login to the system. So if I have 6 phones and all 6 phones want 6 lines, then I will need 36 SIP username/passwords right? And I still won't be able to tell if line 5 is in use or not. What exactly is a hint? How does a hint translate into turn light on on some phones? -Matthew -- Matthew Boehm, IT DirectorCypress Telecommunications [EMAIL PROTECTED] 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. ___ 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
