[Asterisk-Users] Question about Polycom 601 and expansion module.
Olger Merlos V.
olgerm at ekstromcostarica.com
Mon Mar 27 05:21:32 MST 2006
Hi, I have questions about the Polycom 601 and side card....
1) In the side card the lights all time off... But all functions it's ok.
I need help with extension module of polycom... All works fine... But lights
not work.... So... I don't know when any person or extension is busy...
Any ideas?
,
Olger
On 3/27/06 11:34 PM, "asterisk-users-request at lists.digium.com"
<asterisk-users-request at lists.digium.com> wrote:
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> Today's Topics:
>
> 1. Alarm on Unicall (acriollo)
> 2. Ability to put call on hold via manager? (Steve Totaro)
> 3. Re: Receptionist Phones (was 3Com Phones) (Daniel Hazelbaker)
> 4. Call Waiting Issues (Brad Glonka)
> 5. Wanted: Cd-bootable Fedora+Asterisk (Bruce Komito)
> 6. Master.csv Shell Script (Jeremy)
> 7. Re: Ability to put call on hold via manager? (Alberto Sagredo)
> 8. TE 205P/A102 fit in hp dc7600? (JOSE MANUEL CORTES DAVID)
> 9. Re: * Meetme Freeze patch found (Brent Torrenga)
> 10. Re: RE : [Asterisk-Users] Stability of Asterisk with 2 x
> TDM400P cards (6analogue lines) (Krzysztof Drewicz)
> 11. queue caveats (asterisk at anime.net)
> 12. RE: Bluetooth headset in handsfree modewith SJPhoneor X-lite
> (wendell hamilton)
> 13. Re: Config File Management (Giovanni Miano)
> 14. RE: Ability to put call on hold via manager? (Steve Totaro)
> 15. Re: Authorization by ip (Giovanni Miano)
> 16. Re: Call Simulator (Giovanni Miano)
> 17. Re: Alarm on Unicall (Melcon Moraes)
> 18. Re: Receptionist Phones (was 3Com Phones) (Justin Moore)
> 19. Re: Master.csv Shell Script (Mojo with Horan & Company, LLC)
> 20. Re: On site installtion Tech. wanted (Richard Amerman)
> 21. Re: Receptionist Phones (Daniel Hazelbaker)
> 22. FXO without answer supervision (Dan Austin)
> 23. Re: Call Waiting Issues (C F)
> 24. Re: Caller ID length (C F)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 27 Mar 2006 13:06:03 -0600
> From: acriollo <crmeae at gmail.com>
> Subject: [Asterisk-Users] Alarm on Unicall
> To: asterisk-users at lists.digium.com
> Message-ID:
> <e5ff64dc0603271106q3c43b113u17078eeb50deb1ae at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi all,
> any body can tell me why i am receiving this message in my sever ?
>
> I have running * with 10 Digital Lines, but i am receiving a lot of times
> this message .
> Is a software issue or is a hardware issue ?
>
> Regards.
>
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
> Unicall/5 event Alarm
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
> Unicall/5 Alarm masks 0x0000 0x0004
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
> Unicall/5 Alarm No Alarm raised, Yellow Alarm cleared
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
> Unicall/6 event Alarm
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
> Unicall/6 Alarm masks 0x0000 0x0004
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
> Unicall/6 Alarm No Alarm raised, Yellow Alarm cleared
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
> Unicall/7 event Alarm
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
> Unicall/7 Alarm masks 0x0000 0x0004
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
> Unicall/7 Alarm No Alarm raised, Yellow Alarm cleared
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
> Unicall/8 event Alarm
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
> Unicall/8 Alarm masks 0x0000 0x0004
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
> Unicall/8 Alarm No Alarm raised, Yellow Alarm cleared
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
> Unicall/9 event Alarm
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
> Unicall/9 Alarm masks 0x0000 0x0004
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
> Unicall/9 Alarm No Alarm raised, Yellow Alarm cleared
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
> Unicall/10 event Alarm
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
> Unicall/10 Alarm masks 0x0000 0x0004
> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
> Unicall/10 Alarm No Alarm raised, Yellow Alarm cleared
> -------------- next part --------------
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> ------------------------------
>
> Message: 2
> Date: Mon, 27 Mar 2006 14:20:49 -0400
> From: "Steve Totaro" <stotaro at asteriskhelpdesk.com>
> Subject: [Asterisk-Users] Ability to put call on hold via manager?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <DFB93BD730105941BD1A782A1EE9E95CCC07 at 1-0fa9e300af524.asteriskhelpdesk.com>
>
> Content-Type: text/plain; charset="us-ascii"
>
> Does anyone know if there is built in ability to put call on hold via
> the manager interface?
>
> Thanks,
> Steve Totaro
> http://www.asteriskhelpdesk.com
>
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 27 Mar 2006 11:18:10 -0800
> From: Daniel Hazelbaker <daniel at highdesertchurch.com>
> Subject: Re: [Asterisk-Users] Receptionist Phones (was 3Com Phones)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <124F4DD7-FDE4-4156-BCC8-897837AFF38A at highdesertchurch.com>
> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
>
> We may end up using a software solution, but there are two main
> issues with a software solution (for us at least):
>
> 1) For us in particular, our receptionists have ALWAYS (for the past
> 15 years at least) used a physical switchboard style for "routing"
> and seeing availability. From past hardware->software changes we
> know that it will be very frustrating for them. For us, it is much
> more worth it to spend $1,000 to buy each of the two receptionists a
> really nice phone that supports these features rather than get a
> cheap software (though very nice) solution.
>
> 2) Having a software solution can cause grief and frustration to an
> already overworked receptionist. Just a few examples (these are not
> as uncommon as one might think): User quits web browser after
> finishing looking something up on-line, doesn't realize they just
> closed out their switchboard until they need it and it is not there.
> User "gets lost" trying to find the right window while trying to not
> sound like an idiot to the person on the phone. Computer has frozen,
> or otherwise has problems, and must be rebooted.
>
> I do like the look of Asternic, it is very "old-style" and easy to
> get used to, but we would still prefer a hardware solution if
> possible. We may end up having to say, "sorry but you need to deal
> with this for a while until some bugs in the system are resolved
> (i.e. the 7 line problem), but as soon as a hardware solution is
> available we will switch you back to it." Hopefully we can find
> something before we switch, but if not it is good to know that
> software solutions are a viable alternative.
>
>
>> Have you looked that the flash operator panel?
>>
>> http://www.asternic.org/demo.html
>>
>> I know you mentioned not wanting a software solution because of
>> confusion
>> but I think that would be pretty easy to understand.
>>
>> Curt
>>
>> _______________________________________________
>> --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
>>
>>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Mon, 27 Mar 2006 14:18:54 -0500
> From: "Brad Glonka" <glonka at gmail.com>
> Subject: [Asterisk-Users] Call Waiting Issues
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <70d6d74b0603271118w2fec495dhc9091ff1c8e776c7 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> I have two call waiting problems.
>
> I have a POTS line into and FXO port
> and telephones on an FXS port
>
> 1) I can't seem to use the flash button(on the phone) to answer a call
> waiting call.
> I see the callerid coming though and here the call waiting tone,
> but I just can't seem to answer it. The flash button seems to have no
> effect.
>
> I have:
> callwaiting=yes in zapata.conf
>
>
> 2) When the PSTN line is in use and a call comes though via call waiting.
> I don't think it hits my exten => s
> Instead it rings the phone (but as I mentioned above I
> can't seem to answer it)
>
> Thanks for any suggestions.
>
>
> ------------------------------
>
> Message: 5
> Date: Mon, 27 Mar 2006 11:09:45 -0800 (PST)
> From: Bruce Komito <brucek at bagel.com>
> Subject: [Asterisk-Users] Wanted: Cd-bootable Fedora+Asterisk
> To: asterisk-users at lists.digium.com
> Message-ID: <20060327110636.P70224-100000 at mustang.bagel.com>
> Content-Type: TEXT/PLAIN; charset=US-ASCII
>
> I'm in search someone who would be interested in developing a Fedora-baed
> Asterisk system that is bootable from a CD or possible flash. I am aware
> of the various commercial and free solutions out there, but none I have
> found suit our needs...mainly because they are not easily extensible
> and/or upgradeable.
>
> If you are interested in working on such a project, please contact me
> off-list.
>
> Thanks
>
> Bruce Komito
> High Sierra Networks, Inc.
> www.servers-r-us.com
> (775) 236-5815
>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Mon, 27 Mar 2006 14:25:17 -0500
> From: "Jeremy" <thezerogroup at gmail.com>
> Subject: [Asterisk-Users] Master.csv Shell Script
> To: <asterisk-users at lists.digium.com>
> Message-ID: <44283c23.4b9614b5.2e6e.08d4 at mx.gmail.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Im not looking for anything super detailed, just something to run through
> the master.csv file and give total time per account code. . . .does anyone
> out there have a script like this I could work from?
>
>
>
> ------------------------------
>
> Message: 7
> Date: Mon, 27 Mar 2006 21:35:38 +0200
> From: Alberto Sagredo <asagredo at peoplecall.com>
> Subject: Re: [Asterisk-Users] Ability to put call on hold via manager?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <44283E8A.30506 at peoplecall.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> You could park it to parking extensiones.
>
> Does it help you?
>
> Steve Totaro escribió:
>> Does anyone know if there is built in ability to put call on hold via
>> the manager interface?
>>
>> Thanks,
>> Steve Totaro
>> http://www.asteriskhelpdesk.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
>>
>
>
>
> ------------------------------
>
> Message: 8
> Date: Mon, 27 Mar 2006 14:43:35 -0500
> From: "JOSE MANUEL CORTES DAVID" <jmcortes at puj.edu.co>
> Subject: [Asterisk-Users] TE 205P/A102 fit in hp dc7600?
> To: <asterisk-users at lists.digium.com>
> Message-ID:
> <ECCE1C17FAB85140A69454A40B443FCA46251E at CORREOWEB.puj.edu.co>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi
>
> I would like to know if the TE 205 fit in a hp dc7600? what about the A 102
> from Sangoma?
>
> Thanks
>
> Jose Manuel Cortes
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> ------------------------------
>
> Message: 9
> Date: Mon, 27 Mar 2006 13:41:37 -0600
> From: "Brent Torrenga" <lists at torrenga.com>
> Subject: [Asterisk-Users] Re: * Meetme Freeze patch found
> To: <asterisk-users at lists.digium.com>
> Message-ID: <004f01c651d6$76d4da30$7200a8c0 at oscar>
> Content-Type: text/plain; charset="US-ASCII"
>
> Forgoe the patch, just upgrade to 1.2.6. The changelog lists it as a fix
> from 1.2.5 to 1.2.6.
>
>
>> I'm a bit newbie, could you tell me how to i apply the patch?
>>
>> Thanks in advance
>> Marco Mouta
>>
>> On 3/27/06, Benoit Panizzon <benoit.panizzon at imp.ch> wrote:
>>> On Friday 24 March 2006 16:05, Benoit Panizzon wrote:
>>>> Hi all
>>>>
>>>> Apparently there is a patch for those 1.2.4/5 MeetMe Freezes:
>>>>
>>>> http://bugs.digium.com/view.php?id=5884
>>>>
>>>> Haven't tried it out yet.
>>>
>>> I can now confirm: No freezes/crashes anymore since I applied the patch.
>>>
>>> -Benoit-
>
> Sincerely,
>
> Brent A. Torrenga
> brent.torrenga at torrenga.com
>
> Torrenga Engineering, Inc.
> 907 Ridge Road
> Munster, Indiana 46321-1771
>
> 219.836.8918x325 Voice
> 219.836.1138 Facsimile
> www.torrenga.com
>
>
>
> ------------------------------
>
> Message: 10
> Date: Mon, 27 Mar 2006 21:44:53 +0200
> From: Krzysztof Drewicz <drewicz at citicom.pl>
> Subject: Re: RE : [Asterisk-Users] Stability of Asterisk with 2 x
> TDM400P cards (6analogue lines)
> To: f6hqz-m at hamwlan.net, Asterisk Users Mailing List - Non-Commercial
> Discussion <asterisk-users at lists.digium.com>
> Message-ID: <442840B5.2090205 at citicom.pl>
> Content-Type: text/plain; charset=ISO-8859-2
>
> f6hqz-m at hamwlan.net wrote:
>> Hi,
>>
>> Jump to a TDM2402E for 6 POTS lines with hardware echocan.
>> Only one IRQ used, and easy future extensions by adding modules.
>>
>
> Have anyone here used a clone i.e. A1200P-01 (A1200P + 1 FXO100 module) ?
>
>
>
> ------------------------------
>
> Message: 11
> Date: Mon, 27 Mar 2006 11:55:39 -0800 (PST)
> From: asterisk at anime.net
> Subject: [Asterisk-Users] queue caveats
> To: Asterisk-Users at lists.digium.com
> Message-ID: <Pine.LNX.4.63.0603271153340.18525 at sasami.anime.net>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
> According to http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue, under
> the "Notes" section:
>
> "Transfers of calls that are answered out of a queue must be done using
> Asterisk '#' transfers (enabled with the 't' option above). SIP transfers
> result in the Agent remaining affiliated with the call until its eventual
> termination, preventing that agent from being offered another call."
>
> Is this still true in asterisk 1.2.6?
>
> -Dan
>
>
> ------------------------------
>
> Message: 12
> Date: Mon, 27 Mar 2006 11:59:49 -0800
> From: "wendell hamilton" <routerguy at rightsolve.com>
> Subject: RE: [Asterisk-Users] Bluetooth headset in handsfree modewith
> SJPhoneor X-lite
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <88C40680C3714444B9BF371990DE880535877C at mail.rightsolve.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi,
>
> You need to have completely replaced the Microsoft driver, because it
> doesn't support the headset or ctp Bluetooth profiles. This gave me
> fits! I followed the instructions at
> http://www.windowsdevcenter.com/pub/a/windows/2005/07/05/bluetooth.html
> and it works with both a Plantronics and a Motorola Headset, and I can
> answer calls with idefisk, eyebeam, x-lite, and kapanga.
>
> If you end up not having both of these in the Bluetooth service
> selection, you won't end up with the results you're looking for.
>
> HTH
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chuck Bunn
> Sent: Monday, March 27, 2006 9:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Bluetooth headset in handsfree modewith
> SJPhoneor X-lite
>
> Hi,
>
> I am not having trouble with the bluetooth stack since the Toshiba stack
>
> has the headset profile which supports a subset of AT commands
> <http://en.wikipedia.org/wiki/AT_command> from GSM 07.07 for minimal
> controls including the ability to ring, answer a call, hang up and
> adjust the volume. The problem is getting the softphone to work with
> these AT commands so that the answer/hangup function will work from the
> bluetooth headset.
>
> Thanks
>
> wendell hamilton wrote:
>> Try replacing the XP Bluetooth stack with the widcomm drivers...google
>> is your friend!
>>
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chuck
> Bunn
>> Sent: Monday, March 27, 2006 6:21 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [Asterisk-Users] Bluetooth headset in handsfree mode with
>> SJPhoneor X-lite
>>
>> Hi,
>>
>> After much searching I have found that it might be possible to get a
>> bluetooth headset to answer/hangup with SJPhone or Xlite if the
> headset
>> supports handsfree mode. My Toshiba bluetooth stack supports this but
> I
>> have not been able to figure out how to enable it. Also Windows XP
>> desktop bluetooth stack does not support handsfree but Windows CE does
>
>> (go figure). Has anyone got handsfree mode working with a bluetooth
>> headset? How about working with SJPhone or Xlite or some other SIP
>> phone? For some reason the SJPhone when used with a bluetooth headset
>> disconnects/reconnects bluetooth when the answer/hangup button is used
>
>> on the headset (how the hell did that come about). Using a bluetooth
>> headset with a SIP phone and asterisk would really help me by removing
>
>> those pesky wires....
>>
>> Thanks
>> _______________________________________________
>> --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
>>
>> This message is confidential. It may also be privileged or otherwise
> protected by work product immunity or other legal rules. If you have
> received it by mistake, please let us know by e-mail reply and delete it
> from your system; you may not copy this message or disclose its contents
> to anyone. Please send us by fax any message containing deadlines as
> incoming e-mails are not screened for response deadlines. The integrity
> and security of this message cannot be guaranteed on the Internet.
>>
>> _______________________________________________
>> --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
>
>
> ------------------------------
>
> Message: 13
> Date: Mon, 27 Mar 2006 22:00:29 +0200
> From: "Giovanni Miano" <giomiano at gmail.com>
> Subject: Re: [Asterisk-Users] Config File Management
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <d75be1ca0603271200l3536e7cbp9ffe8a269c51b8ca at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> You can use FastAGI
>
> See http://www.asteriskjava.org
>
> 2006/3/27, David Gomillion <dgomillion at eyecarenow.com>:
>>
>> Sorry for thread breaking... I'm on digest.
>>
>>>> I'm curious (ok, well I admit it - it's for perosnal gain) what
>>>> methods people are using to manage asterisk config files when they
>>>> have multiple asterisk systems?
>>>
>>> I'm using CVS. I only have one server right now. I use it on other
>>> clusters to sync files and it works for me..
>>
>> Instead of doing this, I ended up creating a MySQL database and a few
>> scripts to generate the config files for each of my servers. All I have
>> to
>> do is update the database, and the correct server pulls the information
>> from
>> the DB, generates the file, reloads, and sends reboot messages to the
>> proper
>> phones. Very specific to my needs, but extremely fast and effective. And
>> all it requires on each Asterisk server is cron, PHP, and php-mysql.
>>
>> I had to customize a few of the variables inside the PHP scripts for each
>> server, but by putting them close to the top, it's not a real big deal
>> when
>> I update the scripts to customize them for my servers. Mind you, I only
>> have 4 servers on this system, but we don't anticipate growing beyond one
>> more server for a while.
>>
>> One thing to mention that I have found: use lots of macros. Some of my
>> macros require 6 or 7 arguments, but they are extremely flexible and
>> trivial
>> to generate on the fly through these tools. Each extension fits in only
>> one
>> line in the dialplan (calls a macro). Entries in the DB turn on and off
>> features, sets the timeout, forwards to another extension or sends to
>> voicemail, etc.
>>
>> Just what I'm doing. Hope it helps.
>>
>> David
>>
>>
>> _______________________________________________
>> --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
>>
>
>
>
> --
> Giovanni Miano
> -------------- next part --------------
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> ------------------------------
>
> Message: 14
> Date: Mon, 27 Mar 2006 15:10:36 -0400
> From: "Steve Totaro" <stotaro at asteriskhelpdesk.com>
> Subject: RE: [Asterisk-Users] Ability to put call on hold via manager?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <DFB93BD730105941BD1A782A1EE9E95CCC09 at 1-0fa9e300af524.asteriskhelpdesk.com>
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I was thinking about that as an option.
>
> Basically I am integrating a CRM call center app with * and want the agents to
> be able to click a radio button to put callers on hold. They only have analog
> headsets with on-hook and off-hook.
>
> It seems like parking and un-parking the call would be pretty complicated.
>
> Thanks,
> Steve Totaro
>
>
>> -----Original Message-----
>> From: Alberto Sagredo [mailto:asagredo at peoplecall.com]
>> Sent: Monday, March 27, 2006 2:36 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] Ability to put call on hold via manager?
>>
>> You could park it to parking extensiones.
>>
>> Does it help you?
>>
>> Steve Totaro escribió:
>>> Does anyone know if there is built in ability to put call on hold via
>>> the manager interface?
>>>
>>> Thanks,
>>> Steve Totaro
>>> http://www.asteriskhelpdesk.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
>>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ------------------------------
>
> Message: 15
> Date: Mon, 27 Mar 2006 22:02:47 +0200
> From: "Giovanni Miano" <giomiano at gmail.com>
> Subject: Re: [Asterisk-Users] Authorization by ip
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <d75be1ca0603271202i45a61826i8e30288d23535fe3 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> You can use in sip.conf tag "host"
>
> host=192.168.1.1
>
> 2006/3/27, Sam Tam <sam at netenable.co.uk>:
>>
>> Can somebody send me a config of how to authorize SIP client by IP?
>>
>> Sam
>>
>>
>>
>> _______________________________________________
>> --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
>>
>
>
>
> --
> Giovanni Miano
> -------------- next part --------------
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>
> ------------------------------
>
> Message: 16
> Date: Mon, 27 Mar 2006 22:04:52 +0200
> From: "Giovanni Miano" <giomiano at gmail.com>
> Subject: Re: [Asterisk-Users] Call Simulator
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <d75be1ca0603271204s4c6f9cat5d41624c1d0c7635 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> You can use dialout file
>
> http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+auto-dial+out&prev
> iew=20
>
> 2006/3/27, Steve Totaro <stotaro at asteriskhelpdesk.com>:
>>
>> SIPPS is one, I would like to hear of others.
>>
>>
>>
>> Of course you could create a dialplan that loops calls in and out.
>>
>>
>>
>> Thanks,
>> Steve Totaro
>> http://www.asteriskhelpdesk.com
>>
>> ------------------------------
>>
>> *From:* voipman [mailto:emeyeem at gmail.com]
>> *Sent:* Monday, March 27, 2006 6:39 AM
>> *To:* asterisk-users at lists.digium.com
>> *Subject:* [Asterisk-Users] Call Simulator
>>
>>
>>
>> Guyz,
>>
>>
>>
>> I wanna test my asterisk load capability before going to production,
>> anyone know is there any call simulator to test this thing?
>>
>>
>>
>> Thanks in advance,
>>
>>
>>
>> Voipman
>>
>> _______________________________________________
>> --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
>>
>>
>>
>
>
> --
> Giovanni Miano
> -------------- next part --------------
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>
> ------------------------------
>
> Message: 17
> Date: Mon, 27 Mar 2006 17:05:05 -0300
> From: Melcon Moraes <melcon at principaltelecom.com.br>
> Subject: Re: [Asterisk-Users] Alarm on Unicall
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <44284571.4040706 at principaltelecom.com.br>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> What about some unicall.conf and zaptel.conf lines?
>
> []'s
> MM
>
> acriollo wrote:
>> Hi all,
>> any body can tell me why i am receiving this message in my sever ?
>>
>> I have running * with 10 Digital Lines, but i am receiving a lot of
>> times this message .
>> Is a software issue or is a hardware issue ?
>>
>> Regards.
>>
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
>> Unicall/5 event Alarm
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
>> Unicall/5 Alarm masks 0x0000 0x0004
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
>> Unicall/5 Alarm No Alarm raised, Yellow Alarm cleared
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
>> Unicall/6 event Alarm
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
>> Unicall/6 Alarm masks 0x0000 0x0004
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
>> Unicall/6 Alarm No Alarm raised, Yellow Alarm cleared
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
>> Unicall/7 event Alarm
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
>> Unicall/7 Alarm masks 0x0000 0x0004
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
>> Unicall/7 Alarm No Alarm raised, Yellow Alarm cleared
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
>> Unicall/8 event Alarm
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
>> Unicall/8 Alarm masks 0x0000 0x0004
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
>> Unicall/8 Alarm No Alarm raised, Yellow Alarm cleared
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
>> Unicall/9 event Alarm
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
>> Unicall/9 Alarm masks 0x0000 0x0004
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
>> Unicall/9 Alarm No Alarm raised, Yellow Alarm cleared
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:2672 handle_uc_event:
>> Unicall/10 event Alarm
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3091 handle_uc_event:
>> Unicall/10 Alarm masks 0x0000 0x0004
>> Mar 27 12:58:24 WARNING[2586]: chan_unicall.c:3096 handle_uc_event:
>> Unicall/10 Alarm No Alarm raised, Yellow Alarm cleared
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> --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
>
>
> ------------------------------
>
> Message: 18
> Date: Mon, 27 Mar 2006 15:08:16 -0500
> From: "Justin Moore" <wantmoore at gmail.com>
> Subject: Re: [Asterisk-Users] Receptionist Phones (was 3Com Phones)
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <75835fe10603271208v45c1d8b2kbfbffb829bfdab7a at mail.gmail.com>
> Content-Type: text/plain; charset=UTF-8
>
> On 3/27/06, Daniel Hazelbaker <daniel at highdesertchurch.com> wrote:
>> I have seen that the polycom setup (601+sidecar) works but only for up to 7
>> phones
>
>> From what I've seen, each sidecar supports up to 14 additional
> stations. Three of those along with the 5 buttons on the 601 comes up
> to 47 on my calculator. Is there a known problem with the 601+sidecars
> and * that prevents the user from being able to monitor more than 7
> extensions?
>
> Just curious as I've been leaning toward this for our receptionist as
> well (only 12 extensions to monitor...)
>
> --
> Justin Moore
> aka wantmoore
> ---------------------------------------
> www.wantmoore.com
>
> ------------------------------
>
> Message: 19
> Date: Mon, 27 Mar 2006 11:22:21 -0900
> From: "Mojo with Horan & Company, LLC" <mojo at horanappraisals.com>
> Subject: Re: [Asterisk-Users] Master.csv Shell Script
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4428497D.1030907 at horanappraisals.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> If you've got PHP installed, here's one I made for our office:
>
> http://horanappraisals.com/asterisk/total_account_codes/
>
> Run it with no parameters to check Master.csv in the current directory,
> or pass the filename to parse as the first parameter.
>
> # ./total_account_codes /var/log/asterisk/cdr-csv/Master.csv
> "test" total is 310 seconds or 5.17 minutes or 0.09 hours
> "" total is 33130 seconds or 552.17 minutes or 9.2 hours
>
> #
>
> The second line totals all lines with no account code specified.
>
> hth moj
>
> Jeremy wrote:
>> Im not looking for anything super detailed, just something to run through
>> the master.csv file and give total time per account code. . . .does anyone
>> out there have a script like this I could work from?
>>
>> _______________________________________________
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>>
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>>
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