Re: [asterisk-users] Wierd question - Give me your opinion please

[Jared Baxley]

My gut tells me to isolate the analog from the "real" pbx as much as
possible, but in the end it all has to be connected my the lan. I rarely
use FXS devices, so i'm leary of trusting the module to burn and save the
card.

The closest building is 950 ft, the second is 1850 ft. These two buildings
are connected via LRE's using existing 6 Pair, Unfortunately re
cabling isn't an option. Other buildings are even further from the office,
about 15 or so scattered about that only require 1 phone each.

Doe to the flakiness of the power, the fact that they generate and have
solar augmenting the service company, Homeplug and the like are not viable
options. No guaranteei all the buildings are even on the same transformer I
have to work within the confines of existing Telephone cabling between
buildings.

Perhaps one day a strong enough wireless infrastructure can be built to
support these phones.

Now I wonder if I can even get a NEW server with 2 PCI interfaces...


On Tue, Feb 5, 2013 at 2:17 AM, Leandro Dardini  wrote:

> Both the Sangoma and the Digium card are module based, so in the event of
> a electrical shock, the module will burn and the board will be safe.
> About the "too long" lines, the UTP cable is not the only type of cable
> capable of running ethernet. Skipping the too much expensive fiber optics,
> you can find some cheap RG58 cable with BNC connectors. 10BaseT can run up
> to 185 meters and you can have stations in the middle. If the distances are
> longer and you don't like to get your hands dirty with old equipment, then
> you can use the new Homeplug standard, capable of running ethernet on
> almost any electrical media for hundred and hundred of meters of distance.
>
> Leandro
>
>
> 2013/2/5 Carlos Alvarez 
>
>> If you have the budget for two machines, run all services on one and keep
>> the other for a hot backup.  Rsync the configs nightly.  I'm guessing that
>> spare parts/repairs are far away from where you will be?
>>
>>
>> On Mon, Feb 4, 2013 at 9:11 PM, Jared Baxley wrote:
>>
>>> Client - Not for Profit in the Middle of the Jungle/Rain Forrest
>>>
>>> Infrastructure - Datacenter is Non Climate Controlled, Prone to
>>> Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building,
>>> Hodge Podge of DYI wiring across remaining buildings. Phones - Total of
>>> about 50 extensions. Only about 25 - 30 phones will be IP phones, 20-30
>>> more will have to be analog due to the distance.
>>>
>>> Analog Extensions will be on Digium TDM2400 or Sangoma A400 Cards.
>>>
>>> Analog extensions WILL Hit a Surge Gate before the cards, and as much
>>> precaution on grounding protection and power protection is being taken as
>>> possible. The cards WILL BE PCI not PCI-e (They are being donated)
>>>
>>> A New Dell Power-edge Server will be acquired for the PBX
>>>
>>> HERE IS MY QUESTION
>>>
>>> Would you purchase a NEW TOWER Server with PCI slots to accommodate the
>>> cards,
>>>
>>> OR
>>>
>>> Purchase a NEW RACK MOUNT server for the PBX, and Buy/Build a Cheaper
>>> Server just for the analog extensions,
>>>
>>>
>>> I'm torn... The ease of management of one server, or the "isolation" of
>>> analog extensions scattered through the jungle on it's own server.
>>>
>>> Opinions?
>>>
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>>
>>
>>
>> --
>> Carlos Alvarez
>> TelEvolve
>> 602-889-3003
>>
>>
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Re: [asterisk-users] Wierd question - Give me your opinion please

[Jared Baxley]

Minimum 3 days to Next Day Air something in.

My real concern is the Analog Extensions getting hit by lightening and
taking out the server, but since the cards are module based hopefully this
will not happen. I may use your hot spare idea.


On Mon, Feb 4, 2013 at 10:14 PM, Carlos Alvarez wrote:

> If you have the budget for two machines, run all services on one and keep
> the other for a hot backup.  Rsync the configs nightly.  I'm guessing that
> spare parts/repairs are far away from where you will be?
>
>
> On Mon, Feb 4, 2013 at 9:11 PM, Jared Baxley wrote:
>
>> Client - Not for Profit in the Middle of the Jungle/Rain Forrest
>>
>> Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding,
>> and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge
>> Podge of DYI wiring across remaining buildings. Phones - Total of about 50
>> extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will
>> have to be analog due to the distance.
>>
>> Analog Extensions will be on Digium TDM2400 or Sangoma A400 Cards.
>>
>> Analog extensions WILL Hit a Surge Gate before the cards, and as much
>> precaution on grounding protection and power protection is being taken as
>> possible. The cards WILL BE PCI not PCI-e (They are being donated)
>>
>> A New Dell Power-edge Server will be acquired for the PBX
>>
>> HERE IS MY QUESTION
>>
>> Would you purchase a NEW TOWER Server with PCI slots to accommodate the
>> cards,
>>
>> OR
>>
>> Purchase a NEW RACK MOUNT server for the PBX, and Buy/Build a Cheaper
>> Server just for the analog extensions,
>>
>>
>> I'm torn... The ease of management of one server, or the "isolation" of
>> analog extensions scattered through the jungle on it's own server.
>>
>> Opinions?
>>
>> --
>> _
>> -- 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
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Carlos Alvarez
> TelEvolve
> 602-889-3003
>
>
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[asterisk-users] Wierd question - Give me your opinion please

[Jared Baxley]

Client - Not for Profit in the Middle of the Jungle/Rain Forrest

Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding,
and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge
Podge of DYI wiring across remaining buildings. Phones - Total of about 50
extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will
have to be analog due to the distance.

Analog Extensions will be on Digium TDM2400 or Sangoma A400 Cards.

Analog extensions WILL Hit a Surge Gate before the cards, and as much
precaution on grounding protection and power protection is being taken as
possible. The cards WILL BE PCI not PCI-e (They are being donated)

A New Dell Power-edge Server will be acquired for the PBX

HERE IS MY QUESTION

Would you purchase a NEW TOWER Server with PCI slots to accommodate the
cards,

OR

Purchase a NEW RACK MOUNT server for the PBX, and Buy/Build a Cheaper
Server just for the analog extensions,


I'm torn... The ease of management of one server, or the "isolation" of
analog extensions scattered through the jungle on it's own server.

Opinions?
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Re: [asterisk-users] DECT Solution

[Jared Baxley]

I second what Carlos said. The Spectralink is the quality and reliability
you want.
On Jan 24, 2013 2:06 PM, "Carlos Alvarez"  wrote:

> On Thu, Jan 24, 2013 at 12:52 PM, Patrick Lists <
> asterisk-l...@puzzled.xs4all.nl> wrote:
>
>>
>> Polycom also has DECT stuff. I doubt it will come cheap.
>> http://spectralink.polycom.**com/dect_communications/index.**html
>
>
> Not cheap, but this is the solution for large installations (over say 10
> or 20 handsets or big spaces).  Reliable, easy to work with.
>
> --
> Carlos Alvarez
> TelEvolve
> 602-889-3003
>
>
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Re: [asterisk-users] MoH with message on intervals

[Jared Baxley]

Or you could just do a Breakout IVR if they are in a queue ... easy to
manage and update.


On Mon, Jan 21, 2013 at 4:43 PM, Danny Nicholas  wrote:

> The "simplest" way to do it would be to use sox to remix your moh file with
> the message like this:
> Let's say you're using the standard file macroform-cold_day.wav. First you
> split it into two minute segments like so
> Sox macroform-cold_day.wav seg1.wav trim 0.0 120.0
> Sox macroform-cold_day.wav seg2.wav trim 0.0 120.0
> Sox macroform-cold_day.wav seg3.wav trim 0.0 120.0
>
> Now put it back together with your message inserted like this:
> Sox seg1.wav yourmessage.wav seg2.wav yourmessage.wav seg3.wav
> yourmessage.wav macroform-cold_day.wav
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett
> Sent: Monday, January 21, 2013 4:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] MoH with message on intervals
>
> I'm talking to somebody who wants to have a recorded message play
> periodically for people on hold.
>
> An example would be interrupting the hold music every two minutes to play a
> message with business hours and current specials.
>
> Seems like you could fake it by breaking the music files into two minute
> chunks with alphabetical file names, and using sort=alpha. It seems like
> there might also be possible ways to do in the dialplan with
> 'Set(CHANNEL(musicclass)=' or a combination of StartMusicOnHold() and
> StopMusicOnHold().
>
> Can anybody point me in the right direction?
>
>
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Re: [asterisk-users] [OT] Polycom IP450 Firmware Issues

[Jared Baxley]

You probably have to do an incremental upgrade to 3.3.1 or something before
you can load 4.x
On Dec 6, 2012 3:10 PM, "Tim Nelson"  wrote:

> I have a site with Polycom handsets on all the desks, mostly IP650s, some
> IP550s, and some IP450s as well.
>
> I need to update the firmware on the IP450s. However, the firmware simply
> won't load.
>
> The latest firmware (4.0.3 Rev F) supports all phones at this site, and
> was downloaded from here:
> http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
>
> The phone pulls the firmware from the PBX via TFTP (as expected), but
> always results in 'Error: Image is not compatible with the phone'.
>
> As a troubleshooting step, *ALL* firmware has been removed from the TFTP
> root, and *ONLY* the new firmware placed there. So, is the Polycom firmware
> matrix wrong about this phone/firmware compatibility, or am I missing
> something? The bootrom has also been upgraded to the latest without any
> problems.
>
> Thoughts? My head is getting sore from banging it on my desk... :/
>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
>
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Re: [asterisk-users] Need advice on how to implement this ...

[Jared Baxley]

You can park the call, set the timeout low, and have it return to a ring
group.
On Nov 19, 2012 6:15 PM, "Chris Gentle"  wrote:

> I need some advice on how to implement something in my dialplan.
>
> Here's the scenario.  A call comes in on my [incoming] context and I
> answer it.  The call turns out to be for my wife and she needs to answer it
> on a different
> handset somewhere else in the house.
>
> I've tried call parking but the wife acceptance factor is kind of low
> because we don't do it often enough for her to remember how to park and
> unpark.
>
> What I'd really like to do is define an easy DTMF sequence in
> features.conf (like 00) that would send the call back into my [incoming]
> context again,
> just like it was a new incoming call.  Then it could be picked up anywhere
> in the house.
>
> What's the best way to go about this?  I tried doing an AGI script that
> sets context/extension/priority to where I'd like for it to go but it
> doesn't seem to work.
>
> Am I on the right track or is there a better way to do this?
>
> --
> Chris
>
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Re: [asterisk-users] Pager Duty Service on Asterisk

[Jared Baxley]

A google search will yield dozens of how to guides fot asterisk time
conditions. ... but your version and specific deployment must me taken into
account.

If you are looking for someone to implement this for you feel free to
contact me.

Jared Baxley
205.292.0744
On Nov 16, 2012 7:54 PM, "Kaushal Shriyan"  wrote:

>
>
> On Sat, Nov 17, 2012 at 7:22 AM, Jared Baxley wrote:
>
>> You can accomplish this with time conditions.
>>
>
> Thanks Jared. Any docs or tutorials to refer to set up?
>
> Regards,
>
> Kaushal
>
>
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Re: [asterisk-users] Pager Duty Service on Asterisk

[Jared Baxley]

You can accomplish this with time conditions.
On Nov 16, 2012 7:50 PM, "Kaushal Shriyan"  wrote:

> Hi,
>
> Does Asterisk has pager duty feature and write ups or How To's to setup?
>
> Regards,
>
> Kaushal
>
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Re: [asterisk-users] forwarding all calls to cells

[Jared Baxley]

Exactly what Carlos said... properly qualify the clients need, not the
"System they want"
On Nov 9, 2012 1:02 PM, "Noam Birnbaum" 
wrote:

> Yes, the cells could be used as SIP softphones; do you see a benefit to
> doing that?
>
> The reason the client wants to do this is to avoid spending money on
> Ethernet cabling and VoIP desk phones.
>
> noam
>
>
> On Nov 6, 2012, at 4:59 PM, Gerardo Barajas wrote:
>
> Can the cells be used as SIP Softphones?
>
> Saludos/Regards
> --
> Ing. Gerardo Barajas Puente
>
> Proyectos Especiales/Preventa | www.neocenter.com
> T:+52 (55)  8590-9000 x 7003
>
>
> On Tue, Nov 6, 2012 at 6:33 PM, Noam Birnbaum <
> n...@maccentricsolutions.com> wrote:
>
>> Hello everybody,
>>
>> A client wants to install a FreePBX infrastructure, but have all calls
>> forward to their cell phones rather than buying VoIP phones.
>>
>> They would be doing SIP trunks over a Comcast business line.  Probably
>> maximum 6 simultaneous calls.
>>
>> Any gotchas we should warn them about?
>>
>> Thanks!
>>
>> noam
>>
>>
>> Noam Birnbaum
>> El Presidente
>> http://www.desksidemanner.com
>> 415-854-0885 x89
>> tweet @noamb
>>
>>
>>
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Re: [asterisk-users] Critical Asterisk Outage - Installing g729 crashed asterisk.

[Jared Baxley]

Resolved... Thanks to Digium for pointing out I installed a buggy module...
My tired eyes though 1.8.14 was 1.8.1.4 ... Good night all.

On Thu, Oct 18, 2012 at 12:35 AM, Jared Baxley wrote:

> I was following Digium's instructions to the letter to install g729. but
> upon telling asterisk to load the module, the system hung.
>
> after a few minutes later a CTRL-C and attempted to run the command again.
> Same result. any g729 show command returns nothing... no error no results.
>
> Reboot the server and asterisk will not process calls. Freepbx shows the
> following.
>
> [2012-Oct-18 01:21:04] [INFO] (bin/retrieve_conf:109) - found language dir
> fr for directory, not installed on system, skipping
> [2012-Oct-18 01:21:07] [FATAL] (libraries/utility.functions.php:429) -
> retreive_conf failed to get engine information and cannot configure up a
> softwitch with out it. Error: ERROR-UNABLE-TO-PARSE
> [2012-Oct-18 01:21:07] [CRITICAL] (admin/functions.inc.php:366) -
> [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not
> applied
>
> This seems to indicate that the g729 module is working
>
> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: G.729A transcoding
> module version 1.8.0_3.1.5, Copyright (C) 1999-2009 Digium, Inc.
> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This module is supplied
> under a commercial license granted by Digium, Inc.
> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Please see the full
> license text supplied by the accompanying
> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: "register" utility, or
> ask for a copy from Digium.
> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This product includes
> software developed by the OpenSSL Project
> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: for use in the OpenSSL
> Toolkit. (http://www.openssl.org/)
> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Copyright (C) 1998-2006
> The OpenSSL Project
>
> [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered
> action G729LicenseStatus
> [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered
> action G729LicenseList
> [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Host-ID:
> 60:b6:d3:a0:c9:be:6e:46:74:41:07:b3:8e:76:59:b1:ba:5c:59:05
> [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found license
> 'G729-XXX' providing 40 channels
> [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found total of 40
> G.729 licenses
>
> How do i roll this back? Just delete codec_g729a.so ?
>
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Re: [asterisk-users] Critical Asterisk Outage - Installing g729 crashed asterisk.

[Jared Baxley]

Buggy module. I mis-read. and downloaded the one for pre 1.8.4


On Thu, Oct 18, 2012 at 12:53 AM, Steve Totaro <
stot...@totarotechnologies.com> wrote:

>
>
> On Thu, Oct 18, 2012 at 1:49 AM, Steve Totaro <
> stot...@totarotechnologies.com> wrote:
>
>>
>>
>> On Thu, Oct 18, 2012 at 1:35 AM, Jared Baxley wrote:
>>
>>> I was following Digium's instructions to the letter to install g729. but
>>> upon telling asterisk to load the module, the system hung.
>>>
>>> after a few minutes later a CTRL-C and attempted to run the command
>>> again. Same result. any g729 show command returns nothing... no error no
>>> results.
>>>
>>> Reboot the server and asterisk will not process calls. Freepbx shows the
>>> following.
>>>
>>> [2012-Oct-18 01:21:04] [INFO] (bin/retrieve_conf:109) - found language
>>> dir fr for directory, not installed on system, skipping
>>> [2012-Oct-18 01:21:07] [FATAL] (libraries/utility.functions.php:429) -
>>> retreive_conf failed to get engine information and cannot configure up a
>>> softwitch with out it. Error: ERROR-UNABLE-TO-PARSE
>>> [2012-Oct-18 01:21:07] [CRITICAL] (admin/functions.inc.php:366) -
>>> [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not
>>> applied
>>>
>>> This seems to indicate that the g729 module is working
>>>
>>> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: G.729A transcoding
>>> module version 1.8.0_3.1.5, Copyright (C) 1999-2009 Digium, Inc.
>>> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This module is
>>> supplied under a commercial license granted by Digium, Inc.
>>> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Please see the full
>>> license text supplied by the accompanying
>>> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: "register" utility, or
>>> ask for a copy from Digium.
>>> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This product includes
>>> software developed by the OpenSSL Project
>>> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: for use in the OpenSSL
>>> Toolkit. (http://www.openssl.org/)
>>> [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Copyright (C)
>>> 1998-2006 The OpenSSL Project
>>>
>>> [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered
>>> action G729LicenseStatus
>>> [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered
>>> action G729LicenseList
>>> [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Host-ID:
>>> 60:b6:d3:a0:c9:be:6e:46:74:41:07:b3:8e:76:59:b1:ba:5c:59:05
>>> [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found license
>>> 'G729-XXX' providing 40 channels
>>> [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found total of 40
>>> G.729 licenses
>>>
>>> How do i roll this back? Just delete codec_g729a.so ?
>>>
>>>
>> You can do a noload in modules.conf.  This doesn't appear to be the
>> problem though.  It may be.  Did you try saving a change in FreePBX and
>> applying it?
>>
>> It seems more like a FreePBX config error that should be overwritten by
>> FreePBX database to flat files.
>>
>> Thanks,
>> Steve Totaro
>>
>
> See here
> http://www.freepbx.org/forum/freepbx/users/apply-configuration-changes-errors-with-failed-to-get-engine-info-retreive-conf
>
> Very similar problem with FreePBX, your G729 looks fine.
>
> Check permissions and ownership of any files you changed.
>
> Thanks,
> Steve Totaro
>
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[asterisk-users] Hawaii Voip Provider???

[Jared Baxley]

Anybody know a good sip trunking provider for use in Hawaii? Big Island
Specifically. Need to move a client off a dozen pots lines.

Jared
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[asterisk-users] Paetec SIP Trunk

[Jared Baxley]

Has anyone had experience using a SIP trunk provided by Paetec over MPLS?
With or without FreePBX

 

 

 

Regards, 

 

Jared Baxley

 

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Re: [asterisk-users] Two IP addresses and a SIP Trunk

[Jared Baxley]

Many providers ofcer this... if yours do not, find a new sip provider.
On Sep 19, 2012 2:05 PM, "Matej Mailing"  wrote:

> Hi,
>
> we have a failover redundancy with two ISPs and an outside SIP Trunk
> provider for the telephony. Since network failures of our primary ISP
> line happen sometimes I would be interested into a solution where we
> could use both IPs to connect to our telephony provider. I was talking
> to them and they said they need to have our IP entered into their
> system and that it must be only one IP. Therefore when there is a
> failure on our primary ISP, I can call them and they change the IP to
> our secondary IP - but this takes time and our phones don't work in
> the meantime.
>
> Could this be done automatically and if, how? We have two static IPs.
>
> Thanks a lot.
> Matej
>
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[asterisk-users] Asterisk and Wave files.

[Jared Baxley]

I normally do all IVR recordings at an endpoint, but i have a set of voice
prompts i need to use for a specific IVR.

Without converting them on the box, is there any specific utility you all
use for converting sounds to Asterisk friendly files?

what are the acceptable parameters? can audacity or similar make the co
version?

Thanks in advance.
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Re: [asterisk-users] Asterisk on Dynamic IP to a SIP extension

[Jared Baxley]

You were correctly informed about comcast. You should have. No issues
On Jul 28, 2012 8:49 AM, "Paul Belanger" 
wrote:
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Re: [asterisk-users] How to set SIP to auto answer in the dial plan .

[Jared Baxley]

You also have to send the alert info you particular phone needs to make it
autoanswer.
On Jul 13, 2012 4:53 AM, "upendra"  wrote:

> Hi,
>
> thanks , i need to put this in the sip context...
>
> regards
> Upendra.
>
> On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza  > wrote:
>
>> try with SipAddHeader(uri=answer-after=0)
>>
>> check syntax for Addheader
>>
>> Regards,
>> Zohair Raza
>>
>>
>>
>>
>> On Fri, Jul 13, 2012 at 1:42 PM, upendra  wrote:
>> > Hi,
>> >
>> >
>> > I am trying to write dial plan for sip to auto answer (auto attend) the
>> > incoming call to the sip phone.
>> >
>> > - If i call from sip1 to sip2 then sip2 should automatically answer the
>> call
>> > and play some sound file.
>> > I am trying to do this but as new to the asterisk dial plan
>> configuration ,
>> > so not able Todo this.
>> > help me if anyone already done this setup.
>> >
>> >
>> >
>> > Regards
>> > Upendra.
>> >
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>
>
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