[asterisk-users] Can a BLF show busy only if all devices are busy?

2013-08-09 Thread Carlos Alvarez
We all know you can monitor multiple devices in one hint, and it shows busy
if any device in the group is busy.  This is good for a user with multiple
devices, but not useful for teams where any person could take a call, like
a customer service group.  Does anyone know if it's possible to have a hint
with multiple devices which only shows busy if every device is busy?

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Re: [asterisk-users] Can a BLF show busy only if all devices are busy?

2013-08-09 Thread Carlos Alvarez
On Fri, Aug 9, 2013 at 2:32 PM, Alec Davis siva...@paradise.net.nz wrote:

 **
 Sounds like you are using queues, you may be able to use the following;


In the cases I'm thinking of, we aren't using queues.  It's just 2-4 phones
and they don't want a queue specifically, so I've just done a multi-device
dial to keep things simple.  I think I could probably move them to a queue
instead.  Unless someone else has an idea.


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Re: [asterisk-users] limitation on number of contexts in extensions.conf

2013-07-25 Thread Carlos Alvarez
On Wed, Jul 24, 2013 at 11:49 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:

 Hello

 Asterisk version 1.6.2.9.

 I want to know is there any limitation on number of contexts or including
 external file (#include filename) which can be defined in
 extensions.conf. When I try to include around 40 external files, my
 dialplan doen't get reloaded.


There probably is a limit, but I don't know what it is.  We have many
hundreds of contexts and around 80 include files in our main server.  My
guess is you have an error somewhere.  If you show dialplan, does it seem
to stop at a certain point as if it loaded only up to a certain
file/directory?

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Re: [asterisk-users] PoE L3 Switches

2013-07-15 Thread Carlos Alvarez
On Mon, Jul 15, 2013 at 8:40 AM, Eric Wieling ewiel...@nyigc.com wrote:

 Unless it runs IOS, I don't think most of us would consider that box a
 Cisco   Likely it is a Cisco branded switch with Linksys hardware, i.e.
 consumer grade stuff.


They work well in small business.  They have a command line that looks and
feels like IOS, though I have no idea if it is or not.

Note that there is a confirmed bug with LLDP and auto-VLAN on the SF/SG
switches and I haven't heard that they fixed it.  If you have phones with
CDP, or manually provision VLANs, or don't use VLANs, then no problem.
 They work great and are reliable.


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Re: [asterisk-users] Dongle or extra channel and sip SMS

2013-07-15 Thread Carlos Alvarez
Something to check out:

http://www.kickstarter.com/projects/smush/smart-sms-texting-for-everyone-the-smushbox

I'm not affiliated with them at all, but have done business with the
company on other things and have always been happy.



On Mon, Jul 15, 2013 at 1:57 AM, Chris Bagnall
aster...@lists.minotaur.ccwrote:

 On 15/7/13 3:00 am, bilal ghayyad wrote:

 I need to be able to send SMS messages for campaign or for specific
 users, also I need to be able to receive SMS messages and do automatic
 reply.


 In my experience, SMS is something best done out of Asterisk. That's not
 to say that Asterisk can't do it, of course, just that there are providers
 out there who can give you a nice friendly API for easy integration into
 your application. This is especially true if you need to send *lots* of
 messages in a short space of time: simply adding a single mobile device
 with a single SIM isn't going to cut it - you're going to need a bunch of
 them, at least. All of those will likely have different numbers, so you're
 going to have to handle that for receiving messages. Then you have to
 consider that some networks will charge more to send messages to numbers on
 the same network vs. a different network, so you might have to separate out
 your numbers into networks (easy if they've never been ported; more tricky
 if they have).

 Based on past projects (in the UK), the cost of multiple SIM contracts,
 the necessary hardware to connect them, development time, etc., is usually
 more than the cost of paying a third party with a suitable API x per
 message to deliver them on your behalf.

 Kind regards,

 Chris
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[asterisk-users] Is Asternic.net out of business (Flash Operator, Call Center Stats)?

2013-06-17 Thread Carlos Alvarez
We have licensed both products and sent a support request on 6/11, with
zero reply or any activity on it at all so far.  No replies to subsequent
ticket updates or e-mails.


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Re: [asterisk-users] Is Asternic.net out of business (Flash Operator, Call Center Stats)?

2013-06-17 Thread Carlos Alvarez
No vacation notice, nothing, other than the system auto-replying saying
that the ticket will be closed because we didn't have any action on it.
 Rather distressing for our customers.



On Mon, Jun 17, 2013 at 5:22 PM, Gregory Malsack gmals...@coastalacq.comwrote:

 No. Although Nicolas may have gone on holiday. I just purchased 2 licenses
 for fop2 a month or so ago.

 Carlos Alvarez car...@televolve.com wrote:

 We have licensed both products and sent a support request on 6/11, with
 zero reply or any activity on it at all so far.  No replies to subsequent
 ticket updates or e-mails.
 
 
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Re: [asterisk-users] I need a second opinion on a new phone system deployment

2013-06-15 Thread Carlos Alvarez
Interesting product that I was very interested in, but the licensing has
one huge glaring problem.  Be sure to read the FAQ carefully.  If your
hardware fails and you replace almost anything in the machine, you have to
pay for the product again.


On Sat, Jun 15, 2013 at 10:42 AM, Michelle Dupuis mdup...@ocg.ca wrote:

  ...

 For redundant/failover of Asterisk checkout HAAST at www.generationd.com
 The HAAST product sits between Linux and Asterisk, monitors for failures
 etc, and then fails over to another Asterisk box.  It effectively creates a
 low-cost cluster, moving IP's etc to active peer.  It runs with most Linux
 and Asterisk distro's, and avoids the issues of single point of failure.
 etc.

 Michelle
 (generationD)

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Re: [asterisk-users] Polycom and forwarding.

2013-05-15 Thread Carlos Alvarez
On Wed, May 15, 2013 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote:

 Hey, all.  I've got an office set up with Asterisk, and forwarding's got a
 bit of a glitch:
 When they forward, they listen for the remote phone to ring, then hang up.
  If the remote phone doesn't connect, it goes to the original phone's VM.
  Is this Polycom's fault, or Asterisk's?  I've been reading up on
 blind/supervised forwards, and, honestly, have myself more confused than
 when I started.  If someone could give me a solid idea of how forwarding
 works, and a sample of how to send it to a remote extension, and have it
 *not* come back to the original extension, that'd be great.


You said forwarding but described a process that sounds like call
transfer.  I'm going to assume you mean the latter?

We just had a report of this from a customer on their own server.  I
haven't had time to investigate it.  We have confirmed it with Grandstream
and Cisco SPA phones, so it's not just Polycom.

As far as the atxferdropcall someone suggested, I did try that and then the
call is just dropped off into limbo.  The caller is left on hold, and the
nothing happens on the called extension or transfer-to extension.

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Re: [asterisk-users] monitoring Asterisk 1.8

2013-05-09 Thread Carlos Alvarez
Monitor what parts exactly?

Right this moment I'm in the process of installing Munin and the Asterisk
plugin to monitor channel usage, SIP connections, and the like.  The Munin
server is running on a separate machine with just the node software on
Asterisk.



On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote:

 Hello,

 i'm looking for suggestions to monitor Asterisk Server? I installed Nagios
 but no success, I do prefer not to install any web server on the server
 running Asterisk.


 Thanks in advance.
 -Motty

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Re: [asterisk-users] monitoring Asterisk 1.8

2013-05-09 Thread Carlos Alvarez
It's not quick or simple, but there's decent documentation.  I haven't been
saving the links I used, so I can't just give you specific places to look,
other than the best Asterisk plugin:

https://github.com/munin-monitoring/contrib/blob/master/plugins/asterisk/asterisk

TIP:  Use chmod 755 on the plugin files after you install them.

As to installing Munin itself, just start from their web site and get that
running.  You will then install the Asterisk plugin, create an AMI user for
the plugin to connect to, and set the parameters for the plugin to the
server IP and AMI account you just created.

Right now I'm working on being able to monitor the servers without
installing the plugin on the Asterisk box.  This will give Asterisk stats
only, but no server stats.  Again, what specific things do you want to
monitor?



On Thu, May 9, 2013 at 12:53 PM, motty cruz motty.c...@gmail.com wrote:

 Thanks for the suggestion Carlos,

 do you have a HowTo? can you point me to one.

 I unsuccessfully follow one found using google. I'm using CentOs 6.0

 Thanks,
 Motty


 On Thu, May 9, 2013 at 12:38 PM, Carlos Alvarez car...@televolve.comwrote:

 Monitor what parts exactly?

 Right this moment I'm in the process of installing Munin and the Asterisk
 plugin to monitor channel usage, SIP connections, and the like.  The Munin
 server is running on a separate machine with just the node software on
 Asterisk.



 On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote:

 Hello,

 i'm looking for suggestions to monitor Asterisk Server? I installed
 Nagios but no success, I do prefer not to install any web server on the
 server running Asterisk.


 Thanks in advance.
 -Motty

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Re: [asterisk-users] monitoring Asterisk 1.8

2013-05-09 Thread Carlos Alvarez
Then you want a queue manager and reporting tool.  Usually when people say
monitor Asterisk is has to do with the state of the system itself.  You
should look at http://www.asternic.net and similar products.  Munin will
tell you channels in use, but not the other stuff you want.



On Thu, May 9, 2013 at 1:12 PM, motty cruz motty.c...@gmail.com wrote:

 Thanks for your help; I just want to monitor the queue, calls on hold
 average time, incoming out going call, I only want to monitor Asterisk, not
 the server Asterisk in running on.

 thanks,
 -Motty


 On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote:

 http://opennms.org/wiki/Installation:Yum


 On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote:

 I'm using opennms and It's working fine.





 On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote:

 Hello,

 i'm looking for suggestions to monitor Asterisk Server? I installed
 Nagios but no success, I do prefer not to install any web server on the
 server running Asterisk.


 Thanks in advance.
 -Motty

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[asterisk-users] No early media on 302 redirect via two servers

2013-05-08 Thread Carlos Alvarez
We have a situation where we get no early media in this call flow:

VoIP origination provider
Server1 (our server)
Customer server
Customer phone with call-forward set
Server1 to dial the forward-to number

Then there is no early media while the forward-to number is ringing.  Our
server is Asterisk 1.6 and theirs is 1.8.

I tried promiscredir=yes and then the calls fail altogether because rather
than using the local channel, it makes a SIP call that is not allowed.  I
don't think that the FORWARD_CONTEXT variable is used in these versions,
because setting it doesn't impact the channel selection at all.

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Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider

2013-05-08 Thread Carlos Alvarez
On Tue, May 7, 2013 at 10:05 PM, Satish Barot satish4aster...@gmail.comwrote:



 promiscredir= yes in sip.conf should help you achieve your requirement.


I haven't been able to get that to work in a similar situation, except we
are the provider.  It results in the new invite being from the CLID of the
original caller, and fails.


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[asterisk-users] Playing a sound file during a call

2013-05-02 Thread Carlos Alvarez
I have a customer who would like to play a series of sound files
during a phone call on demand.  There would be several played in order
during a call.  Any simple ideas on doing that without developing a
whole web app to do it via AMI?

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Re: [asterisk-users] Playing a sound file during a call

2013-05-02 Thread Carlos Alvarez
Unfortunately that only plays the file to one side according to the
examples, so there's no way for the other person to know when it's
done.  The caller on the Asterisk server would start the playback, and
would need to know when it's done.

On Thu, May 2, 2013 at 2:58 PM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
 I think features.conf has what you want under the [applicationmap] setting.
 They even have an example that would be almost exactly like what you want.
 From the example:

 ;testfeature = #9,peer,Playback,tt-monkeys  ;Allow both the caller and
 callee to play
 ;;tt-monkeys to the opposite
 channel

 Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



 From:Carlos Alvarez car...@televolve.com
 To:Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com,
 Date:05/02/2013 04:53 PM
 Subject:[asterisk-users] Playing a sound file during a call
 Sent by:asterisk-users-boun...@lists.digium.com
 



 I have a customer who would like to play a series of sound files
 during a phone call on demand.  There would be several played in order
 during a call.  Any simple ideas on doing that without developing a
 whole web app to do it via AMI?

 --
 Carlos Alvarez
 TelEvolve
 602-889-3003

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Re: [asterisk-users] Playing a sound file during a call

2013-05-02 Thread Carlos Alvarez
Good point, and that should be usable for the customer.

However I'm finding that I can only have about 25 files available to
play, and they need 30.  Still trying to figure out why that would be,
seems like the set command can't parse all 30 possible feature names.

On Thu, May 2, 2013 at 3:07 PM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
 Add MOH_Class onto the example and the idle channel will hear music on hold
 until the playback is complete on the other channel.

 Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



 From:Carlos Alvarez car...@televolve.com
 To:Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com,
 Date:05/02/2013 05:02 PM
 Subject:Re: [asterisk-users] Playing a sound file during a call
 Sent by:asterisk-users-boun...@lists.digium.com
 



 Unfortunately that only plays the file to one side according to the
 examples, so there's no way for the other person to know when it's
 done.  The caller on the Asterisk server would start the playback, and
 would need to know when it's done.

 On Thu, May 2, 2013 at 2:58 PM, Kevin Larsen
 kevin.lar...@pioneerballoon.com wrote:
 I think features.conf has what you want under the [applicationmap]
 setting.
 They even have an example that would be almost exactly like what you want.
 From the example:

 ;testfeature = #9,peer,Playback,tt-monkeys  ;Allow both the caller and
 callee to play
 ;;tt-monkeys to the opposite
 channel

 Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



 From:Carlos Alvarez car...@televolve.com
 To:Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com,
 Date:05/02/2013 04:53 PM
 Subject:[asterisk-users] Playing a sound file during a call
 Sent by:asterisk-users-boun...@lists.digium.com
 



 I have a customer who would like to play a series of sound files
 during a phone call on demand.  There would be several played in order
 during a call.  Any simple ideas on doing that without developing a
 whole web app to do it via AMI?

 --
 Carlos Alvarez
 TelEvolve
 602-889-3003

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Playing a sound file during a call

2013-05-02 Thread Carlos Alvarez
In case anyone else sees this discussion in the future, the
Set(__DYNAMIC_FEATURES) line can't be over a certain length or it
stops parsing anything after that.

Thanks for the tips, Kevin.


On Thu, May 2, 2013 at 3:37 PM, Carlos Alvarez car...@televolve.com wrote:
 Good point, and that should be usable for the customer.

 However I'm finding that I can only have about 25 files available to
 play, and they need 30.  Still trying to figure out why that would be,
 seems like the set command can't parse all 30 possible feature names.

 On Thu, May 2, 2013 at 3:07 PM, Kevin Larsen
 kevin.lar...@pioneerballoon.com wrote:
 Add MOH_Class onto the example and the idle channel will hear music on hold
 until the playback is complete on the other channel.

 Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



 From:Carlos Alvarez car...@televolve.com
 To:Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com,
 Date:05/02/2013 05:02 PM
 Subject:Re: [asterisk-users] Playing a sound file during a call
 Sent by:asterisk-users-boun...@lists.digium.com
 



 Unfortunately that only plays the file to one side according to the
 examples, so there's no way for the other person to know when it's
 done.  The caller on the Asterisk server would start the playback, and
 would need to know when it's done.

 On Thu, May 2, 2013 at 2:58 PM, Kevin Larsen
 kevin.lar...@pioneerballoon.com wrote:
 I think features.conf has what you want under the [applicationmap]
 setting.
 They even have an example that would be almost exactly like what you want.
 From the example:

 ;testfeature = #9,peer,Playback,tt-monkeys  ;Allow both the caller and
 callee to play
 ;;tt-monkeys to the opposite
 channel

 Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



 From:Carlos Alvarez car...@televolve.com
 To:Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com,
 Date:05/02/2013 04:53 PM
 Subject:[asterisk-users] Playing a sound file during a call
 Sent by:asterisk-users-boun...@lists.digium.com
 



 I have a customer who would like to play a series of sound files
 during a phone call on demand.  There would be several played in order
 during a call.  Any simple ideas on doing that without developing a
 whole web app to do it via AMI?

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Re: [asterisk-users] Playing a sound file during a call

2013-05-02 Thread Carlos Alvarez
On Thu, May 2, 2013 at 5:24 PM, Richard Mudgett rmudg...@digium.com wrote:

 You can also put dynamic feature group names into the DYNAMIC_FEATURES list.

I expected that to have the same limitation, but it doesn't.  Works
fine, thanks!


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Re: [asterisk-users] multiple provider for incoming

2013-04-30 Thread Carlos Alvarez
Toll free can be moved by a RESPORG in minutes. I know a company who is
Asterisk based and does high availability for critical numbers. Let me know
if you want an intro.

Sent from my iPhone

On Apr 30, 2013, at 9:09 PM, Matt Hamilton mistral9...@hotmail.com wrote:

 Don,

Inbound reliability is very important. We don't use toll-free numbers, but
we will look into that. I thought porting numbers - not sure about
toll-free though - from one provider to the other took days (not
technically, but paperwork, etc.)

Thanks,
Matt

--
From: d...@donkelly.biz
To: asterisk-users@lists.digium.com
Date: Tue, 30 Apr 2013 22:38:44 -0500
Subject: Re: [asterisk-users] multiple provider for incoming

If inbound reliability is important, you may be able to accomplish what you
want with redundant servers, multiple sip providers and toll-free numbers
that can be readily switched between the sip providers.

--Don

*From:* asterisk-users-boun...@lists.digium.com [
mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com]
*On Behalf Of *Matt Hamilton
*Sent:* Tuesday, April 30, 2013 10:25 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] multiple provider for incoming



The process will depend on your provider, of course, but I know some have
an option that if they are unable to reach
your box, then they can auto-forward to another DID, or to a voicemail
box, or to a user-defined function, etc.

Forwarding to another DID will/should work for us assuming they are going
to be able to do that during a failure on their side. During a recent
outage (I think they had some major issues at one of their switches), they
were not able to send the calls to our box which was online.

Thanks,
Matt

--

Date: Tue, 30 Apr 2013 20:38:19 -0500
From: wcse...@selbytech.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] multiple provider for incoming

On Tue, Apr 30, 2013 at 7:50 PM, David Wessell da...@ringfree.biz wrote:

Hi Matt,



You can't have multiple providers for inbound traffic. You can have
multiple providers for outbound traffic though.



Thanks

David





David,



I'm not sure where you got this information, but it's not accurate.  I've
had multiple inbound and outbound SIP providers for years going to a single
box.  You just get a separate DID from each provider.

Matt,

The process will depend on your provider, of course, but I know some have
an option that if they are unable to reach your box, then they can
auto-forward to another DID, or to a voicemail box, or to a user-defined
function, etc.


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--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com


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Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones

2013-04-29 Thread Carlos Alvarez
Well the solution turned out to be putting the Asterisk server name in
the Proxy field as well as in the server field.  Then it properly
formatted the SIP registration request.

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[asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones

2013-04-28 Thread Carlos Alvarez
We have a new customer with a lot of old phones like the 9133i.  They
won't register, and we see some very strange behavior with them.  If
the SIP peer exists, they simply fail silently, with no error in the
CLI or the messages log.  Nothing works, but no errors.

If the peer does not exist, it's clear that it's registering improperly:

[2013-04-28 13:34:31] NOTICE[3058] chan_sip.c: Registration from
'abc123 sip:abc123@' failed for '68.2.x.x' - No matching peer found

Typically of course we'd expect to see:  sip:abc123@server

We're running the latest available firmware, but it's from 2009.  Any
ideas on this before we just trash all the older phones?

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[asterisk-users] Logging SIP connection status for review

2013-04-10 Thread Carlos Alvarez
Is anyone using something to log SIP results (connected/not, latency) that
they really like?  We do some logging using simple scripts writing the
results of sip show peers to a text file if customers report issues, but it
would be nice to have a tool that logs all the time and lets us do some
better reporting.  For example, graphs of latency in a time range, or a
list of unreachable phones within a range, etc.


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Re: [asterisk-users] Logging SIP connection status for review

2013-04-10 Thread Carlos Alvarez
On Wed, Apr 10, 2013 at 11:02 AM, Steve Edwards
asterisk@sedwards.comwrote:


 dumpcap can capture all of the SIP (and RTP) packets into a series of
 files without a huge performance hit.

 A cron job can pbzip2 the files and delete if over x days old.


That's completely different.  We already run a good packet capture system.
 What I want to see is SIP registration statuses and latency logged about
once a minute.  We do that now by doing a 'sip show peers like x' and
putting it in a text file.  I can then correlate issues with times of high
latency or unreachable phones.  I'd just like to see more reporting and the
ability to correlate times and such.

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Re: [asterisk-users] To queue or not to queue...

2013-03-28 Thread Carlos Alvarez
On Thu, Mar 28, 2013 at 12:55 PM, Gregory Malsack
gmals...@coastalacq.comwrote:

 Here's the scenario~
 150 agents, all are commission based sales reps. 99% of the calls are
 answered within the first ring. the rest are answered between the second
 and third ring. Never in my 4 months with the company has a queue call been
 in the queue more then 20 seconds.

 Problem~
 Several times a week or sometimes a day, the reps will tell me that the
 same call will be answered by 3 or 4 or 5 reps, and none of them get the
 inbound audio. Asterisk only shows 1 of the reps actually connecting the
 call, however the call logs in Eyebeam for all 5 reps, show that they took
 the call and were connected for a short period of time before disconnecting
 the call because there is no inbound audio.


Which version of Asterisk?  Have you looked for solutions to the root
problem?  I don't run any servers with that many agents, but have never run
into issues like this with a few dozen.

Large ring groups can become unwieldy and problematic themselves.  There's
also a limit to how long the entire dial string can be, though I can't
remember what that size is.

You said everything is on a LAN, but have you looked at the possibility of
issues between switches?  Can you examine the logs of bad calls and see if
the failures happen on a specific switch in the network, or other
correlation like that?  Do you use VLANs or layer 3 switching?

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[asterisk-users] Cisco SPA 5xx/3xx/9xx phones don't respond to SIPAddHeader(Call-Info: answer-after=0)

2013-03-21 Thread Carlos Alvarez
All other phones we work with will auto-answer when we do this:

[macro-paging1way]
exten = s,1,SIPAddHeader(Call-Info: answer-after=0)
exten = s,n,Page(${PAGINGLIST})
exten = s,n, Hangup

The SPA phones simply ring.  I have verified that Auto Answer Page is set
to yes (the default).  We've tried a variety of firmware versions and phone
ages, going back to an old 942 and new 504s.  Any ideas?

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Re: [asterisk-users] Cisco SPA 5xx/3xx/9xx phones don't respond to SIPAddHeader(Call-Info: answer-after=0)

2013-03-21 Thread Carlos Alvarez
On Thu, Mar 21, 2013 at 11:58 AM, Optical Phoenix
opticalphoe...@gmail.comwrote:



 Hi Carlos,
 According to this site,
 http://community.linksys.com/t5/VoIP-Phones/SPA942-auto-answer-page-intercom-beeping-loudly/td-p/215064the
  sip string should be 
 Call-Info:\;answer-after=0. I have not tested this yet however.


Thanks, that does work.  Seems to not interfere with the Grandstream and
Polycom phones' operation either.

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Re: [asterisk-users] asterisk with 1000 extensions

2013-03-07 Thread Carlos Alvarez
On Thu, Mar 7, 2013 at 1:44 AM, Duncan Turnbull dun...@e-simple.co.nzwrote:

 This is not school assignment or home work :)  We need to setup in society
 buildings. Each flat will have SIP extension (hard phone) registered on
 asterisk server. Calling between SIP extensions is required. No PSTN /
 ITSP SIP trunking. Just like inter-com feature.

 One way is to install 1000 IP Phones one at each flat
 Secondly, install multiple-line SIP gateways with RJ-11 cabling.

 Is there any other low budget solution for this setup?


Grandstream makes some inexpensive phones that are still very good.

Cheapest hasn't been defined yet.  What's the budget?  Is there existing
networking at these locations?  Will you need switches?  PoE?

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Re: [asterisk-users] Error to install Asterisk

2013-03-06 Thread Carlos Alvarez
I'm going to make an observation here that may upset you, and I don't mean
it to, but it's fact.  If you are so unfamiliar with Linux, you will have a
bad time managing Asterisk servers.  You really need to know how to use the
OS before you can learn to manage services running on it.  I strongly
suggest one of the all-in-one Asterisk variants like AsteriskNOW.  There is
simply no way to run a production server without having to do systems
management regularly.


On Wed, Mar 6, 2013 at 3:01 AM, termo termosel fermit...@hotmail.comwrote:

 Hi,

 this is the outpu to df -h command:

 root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h
 S.ficherosTam.  Usado Disp. % Uso Montado en
 /cow   14G  4,5G  8,7G  34% /
 udev  999M  4,0K  999M   1% /dev
 tmpfs 403M  860K  402M   1% /run
 /dev/sdb1 799M  693M  106M  87% /cdrom
 /dev/loop0668M  668M 0 100% /rofs
 tmpfs1006M   44K 1006M   1% /tmp
 none  5,0M 0  5,0M   0% /run/lock
 none 1006M  100K 1006M   1% /run/shm

 Jordi

 --
 From: fermit...@hotmail.com
 To: asterisk-users@lists.digium.com
 Date: Tue, 5 Mar 2013 17:40:32 +

 Subject: Re: [asterisk-users] Error to install Asterisk

 Hi,

 Ok, tomorrow I will send the output when I will be in the office!

 Thanks!

  From: asterisk_l...@earthshod.co.uk
  To: asterisk-users@lists.digium.com
  Date: Tue, 5 Mar 2013 16:11:01 +
  Subject: Re: [asterisk-users] Error to install Asterisk
 
  On Tuesday 05 March 2013, termo termosel wrote:
   Hi,
   when I try to install Asterisk 11.2.1 the console return error which it
   tells: /usr/bin/ld: final link failed: No space left on device
   and the process exits installation.
   How can I solve this problem? Tmp folder is empty.
   Thanks,Jordi
 
  Try entering this command:
  # df -h
  and paste the complete output in a message.
 
  This will show the amount of space used and remaining on all
 filesystems, in
  human-readable notation (i.e. it will automatically select the units:
 bytes,
  kilo, mega, giga or terabytes, so as to get a sensible figure).
 
  You'll almost certainly have to move some files out of the way. Have you
 got,
  or can you get, a USB external HDD; which either already has a Linux
 ext4 file
  system on it, or contains only sacrificial data?
 
  --
  AJS
 
  Answers come *after* questions.
 
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Re: [asterisk-users] Error to install Asterisk

2013-03-06 Thread Carlos Alvarez
On Wed, Mar 6, 2013 at 10:02 AM, Gertjan Baarda gertjan.baa...@gmail.comwrote:

 Couldn't agree more, Carlos. But then again, haven't we all started this
 way? ;-) The best way to understand Linux is learning the hard way. After
 all, it takes a genius to understand the simplicity of Linux.


If you're going to learn Linux, then learn it, not via some service running
on it.  It's clear in context that the original poster believes that he can
install and run Asterisk without knowing the OS.  This is obviously not
true.  If it's going to be someone's production server, that is scary.  It
also has led to many ASTERISK SUCKS! discussions I've had because there
were problems at the OS level that made the Asterisk server unreliable.

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Re: [asterisk-users] What would cause a drop between two asterisk systems?

2013-03-05 Thread Carlos Alvarez
On Tue, Mar 5, 2013 at 2:32 PM, Hose hose+aster...@bluemaggottowel.comwrote:

 We have an asterisk frontend terminating all our SIP phones to, and an
 asterisk backend with a wildcard PRI card in it connecting to the PTSN.
 The frontend handles 99% of dialplan logic and just hands off anything
 outgoing to the backend via IAX2, which dials out on one of the open
 channels.


IAX is buggy.  We've never seen a reliable system using it.  We've given up
on it.  I'd try SIP.  Easy to do, no real reason not to.

Check all of the networking involved.  Leave a ping test running between
the systems constantly, then see if it dropped packets when you get a
dropped call.


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Re: [asterisk-users] Dialplan / check / tool

2013-02-18 Thread Carlos Alvarez
On Mon, Feb 18, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com wrote:


 A normal dialplan reload command would echo no warning or something
 similair.


The duplicated extension will cause an error.  Something like cannot add
extension in line X because it already exists.

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Re: [asterisk-users] Asterisk calls between 2 private networks

2013-02-07 Thread Carlos Alvarez
On Thu, Feb 7, 2013 at 10:26 AM, Frank fr...@efirehouse.com wrote:

 AJS,

 That is a solution that I am envisaging.
 But I would really love to try to work out with my issue first. It will
 allow me to deploy more phones in separates buildlings in the future. If I
 do the IAX solution, it means that for every building, I need a box.. Which
 I would like to prevent.


Adding more points of failure and more devices to maintain without any real
benefit is always the wrong thing to do.  IAX is also flaky as hell.

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

2013-02-05 Thread Carlos Alvarez
On Tue, Feb 5, 2013 at 11:43 AM, Ira i...@extrasensory.com wrote:

 Personally I think I'd purchase a number of used servers so if one dies,
 you have a backup or three. 25 lines does not need a modern processor and
 it seems silly to spend money for a warranty when you could spend less and
 have a number of spares.


I completely agree with this.  I'd buy several refurb/used machines rather
than one new.

We buy all of our HP refurb servers from these guys:
http://www.nautilusnet.com/
alb...@nautilusnet.com

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Re: [asterisk-users] How to connect a POTS robo alert dialer to asterisk with email notification

2013-02-04 Thread Carlos Alvarez
On 3/2/13 4:59 pm, David Smiley wrote:

  I finally found the perfect solution for me:http://www.amazon.com/La-**
 Crosse-D111-101-E1-WGB-**Wireless-Monitor/dp/**
 B0081UR76G/ref=dp_ob_title_defhttp://www.amazon.com/La-Crosse-D111-101-E1-WGB-Wireless-Monitor/dp/B0081UR76G/ref=dp_ob_title_def
 The device is $69, plus $10/month for alerts.  And I get to monitor the
 temperature online, which is a great bonus.


I use these devices to monitor everything from server rooms to my home
freezer and cigar humidor (they also have monitors that do humidity). This
is the best solution to the problem, not a voice system.


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

2013-02-04 Thread 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 jared.bax...@gmail.com 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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Re: [asterisk-users] send record file to email

2013-02-01 Thread Carlos Alvarez
On Fri, Feb 1, 2013 at 8:35 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote:

 On Friday 01 February 2013, Bob Kyeyune wrote:
  Hello;
  how do i embed and send the recorded file to email automagically
 
  exten = _1XXX,3,MixMonitor(${CALLFILENAME}|b|/usr/sbin/wav2mp3
  ${CALLFILENAME} ${peeremail} ${EXTEN}
 ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
  )
 



Here's an example of how we record and e-mail.  You will need to download
the sendEmail binary.

[conference-record]
exten = s,1,Answer
exten =
s,n,Set(MEETME_RECORDINGFILE=/var/spool/asterisk/monitor/TelEvolve-conf-${UNIQUEID})
exten = s,n,Set(MEETME_RECORDINGFORMAT=wav49)
exten = s,n,Wait(1)
exten = s,n,MeetMe(televolve,cDr)
exten = h,1,System(/usr/sbin/sendEmail -t car...@televolve.com -f
notificati...@televolve.com -u Conference call recording -m Conference
call ${UNIQUEID})


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Re: [asterisk-users] sip register peer (the quest for near 100% availability)

2013-01-31 Thread Carlos Alvarez
On Thu, Jan 31, 2013 at 9:26 AM, Adam Moffett adamli...@plexicomm.netwrote:


 Maybe it's possible to send a NOTIFY to a peer on the last IP it was seen
 at?  I don't think I've seen anything that has a register command, but
 lots of devices can get a check your config or reboot command via SIP
 NOTIFY.


If you can notify, you can call.  This fixes nothing other than refreshing
NAT if that's involved.


 I'm more wondering why the peer is unregistered but we still expect to
 communicate with it.  Other than a network problem or the device being
 unplugged...neither of which could be fixed from the server.


I have a feeling that some people in this discussion have a lack of
understanding about the SIP protocol and the underlying networking that
could affect it.  The original post failed to say whether this was on a LAN
without routing, on a LAN with routing, or a WAN.  Each of those could
result in totally different results and solutions.


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Re: [asterisk-users] (SOLVED) Call parking in a multi-tenant system

2013-01-31 Thread Carlos Alvarez
I figured I'd follow up on this in case anyone else cares.  The
documentation is simply awful and it took a lot of experimenting to make it
work.

In features.conf for each company:

[parkinglot_televolve]
parkext = 700
parkpos = 701-720
context = televolve#parking
parkinghints = yes
parkingtime = 75
courtesytone = beep
parkedplay = both
findslot = next


Include the context declared above in extensions.conf under the dial
context.
include = televolve#parking


In the defaults for the customer's sip.conf file add:
parkinglot=parkinglot_televolve


NOTE:  The parkinglot_ in the name is required!


On Tue, Jan 15, 2013 at 2:08 PM, Bakko asannu...@gmail.com wrote:

  Hello,

 from 1.6.2 version, Asterisk suport multi-tenant parking

 Look at features.conf for a example.

 Regards


 El 15/01/2013 15:58, Carlos Alvarez escribió:

 We use Asterisk as a hosted PBX.  We've had a couple of requests for
 parking, but none of the documentation shows any way to make it aware of
 contexts or otherwise make it multi-tenant.  Have I missed something and
 does anyone know how to make this work?  Would be on Asterisk 1.6 for now,
 1.8 some time soon.

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Re: [asterisk-users] sip register peer (the quest for near 100% availability)

2013-01-30 Thread Carlos Alvarez
On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote:

 Thanks - I was hoping there was some silver bullet to use out there. Thanks
 anyway.


There is.  If you build a reliable network, the phones will simply never
have a problem.  We've got customers with phones that have never lost
contact for years.  Re-registering is just a crutch for a network defect.

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[asterisk-users] Quoting error with gotoiftime

2013-01-25 Thread Carlos Alvarez
I'm getting the following error, and none of us can figure out why:

 [Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror():  syntax error: syntax error, unexpected 'token', expecting
$end; Input:
 = 
  ^


Here is the code that generates it:

[scottsdale#queues-account]
exten = s,1,GotoIfTime(8:00-16:55,mon-fri,*,*?queue)
exten = s,n,Goto(scottsdale#queues-closed,s,1)
exten = s,n(queue),ExecIf($[${prefix} =
]?Queue(azenglish):Queue(azspanish))
exten = s,n(callback),Playback(scottsdale-q/${prefix}callbackmessage)
exten = s,n,Voicemail(@scottsdale,s)
exten = s,n,Hangup


Here is the rest of the call progress surrounding it, which seems to be
working anyway:

-- Executing [2@scottsdale#queues-aax:1]
Goto(SIP/televolve-1-1c7d, scottsdale#queues-account,s,1) in new
stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing [s@scottsdale#queues-account:1]
GotoIfTime(SIP/televolve-1-1c7d, 8:00-16:55,mon-fri,*,*?queue) in
new stack
-- Goto (scottsdale#queues-account,s,3)
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror():  syntax error: syntax error, unexpected 'token', expecting
$end; Input:
 = 
  ^
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [s@scottsdale#queues-account:3]
ExecIf(SIP/televolve-1-1c7d, ?Queue(azenglish):Queue(azspanish))
in new stack


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Re: [asterisk-users] Quoting error with gotoiftime

2013-01-25 Thread Carlos Alvarez
On Fri, Jan 25, 2013 at 9:20 AM, Eric Wieling ewiel...@nyigc.com wrote:

 Looks to me like ${prefix} contains nothing but two quotes.


Which is as it should be unless they choose the Spanish option, but yeah,
maybe that's what is choking Asterisk.

We do this:
exten = _X.,n,Set(prefix=) ;Initialize variable used in
scottsdale#queues-aax

Then in the AAX if they choose 5:
exten = 5,1,Set(prefix=s-); SPANISH

That way we just carry along their preference for Spanish throughout the
system.  Perhaps I need to initialize the variable as e-, but then would
have to rename everything.


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Re: [asterisk-users] Quoting error with gotoiftime

2013-01-25 Thread Carlos Alvarez
On Fri, Jan 25, 2013 at 9:22 AM, Eric Wieling ewiel...@nyigc.com wrote:

 What version does the error occur on?  I suspect more recent versions of
 Asterisk removes extraneous quotes.


This is in 1.8.

Danny's test does support your theory.  It looks like the var is being set
as the quotes, rather than empty.

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Re: [asterisk-users] Quoting error with gotoiftime

2013-01-25 Thread Carlos Alvarez
On Fri, Jan 25, 2013 at 9:31 AM, Danny Nicholas da...@debsinc.com wrote:

 Where possible you should have a VM to try these things as needed.  Where
 not, it isn’t too difficult to duplicate the contexts and do something like
 this

 [default]


I do have a test VM, but I also have a maintenance window for this customer
later tonight for other things, so I was being lazy.  Poor excuse.

Tested it, and it works, thanks guys!

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Re: [asterisk-users] How to assign the button on the IP Phone to a feature?

2013-01-24 Thread Carlos Alvarez
On Thu, Jan 24, 2013 at 8:03 AM, bilal ghayyad bilmar...@yahoo.com wrote:

 Dear;

 Using Cisco IP Phones: How I can assign a button for a function. For
 example, if we pressed on this button, then we need to pickup the call from
 the group.


Which model line?  The SPA series, or the 7900 and similar?  They are
completely different.

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Re: [asterisk-users] DECT Solution

2013-01-24 Thread Carlos Alvarez
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.**htmlhttp://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.

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Re: [asterisk-users] Is there a need to secure RTP ports?

2013-01-23 Thread Carlos Alvarez
On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus s...@open-t.co.uk wrote:

 I have an Asterisk server with one SIP trunk to a SIP provider. As my
 server registers with the SIP provider, I don't have any SIP ports open at
 my end to the Internet. However, I have the RTP ports open (as SIP has some
 trouble with my NAT). My question is - what are the vulnerabilities in this
 scenario at my end? I suppose some man-in-the-middle or eavesdropping
  attack is always a possibility - but that aside, is there anything that
 will attack RTP ports on Asterisk when there are no SIP ports open? I was
 looking into installing fail2ban - until I realised that there is no SIP
 port exposed for an attacker to poke at.


I've been working in IP telephony for about ten years.  I've never once
heard of any attack on the RTP ports.  While you can never say anything is
impossible there's simply nothing listening on those ports.  It's
probably possible to have a DOS attack where someone starts sending RTP to
all of your ports and they would interfere with a call, but they couldn't
do more than that.  That could work if your router has full cone NAT and a
lot of other things fall into place.  Still kind of out there as a real
threat.


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Re: [asterisk-users] Asterisk, Digium phones, and voicemail.

2013-01-22 Thread Carlos Alvarez
On Tue, Jan 22, 2013 at 1:48 PM, Frank fr...@efirehouse.com wrote:

 That worked, thank you.

 Is there a way to program the keys of the Digiums D70 from asterisk ?
 Or does everything needs to be done on the phone itself ?


The whole point of the Digium phones is their tight integration with
Asterisk.  You need the Digium phone module, and you really need to read
the documentation as this is well covered.

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Re: [asterisk-users] Asterisk voicemail minimum length / silence settings

2013-01-22 Thread Carlos Alvarez
On Tue, Jan 22, 2013 at 4:22 PM, asterisk users ast4...@gmail.com wrote:


 What are the right settings for this situation?


We've used the following settings system-wide for about nine years without
one complaint or known issue:

[general]
format = wav49|pcm
maxsecs = 360
minsecs = 4
skipms = 3000
maxsilence = 3
maxlogins = 3
maxgreet = 120
maxmsg = 50
silencethreshold = 128
operator = yes
sendvoicemail = yes
usedirectory = yes
forcename = yes
forcegreeting = yes
saycid = no
review = yes
nextaftercmd = yes
tempgreetwarn = yes

Some of these things may be deprecated depending on the version you're
using.

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Re: [asterisk-users] Asterisk voicemail minimum length / silence settings

2013-01-22 Thread Carlos Alvarez
On Tue, Jan 22, 2013 at 4:43 PM, Don Kelly d...@donkelly.biz wrote:

 But with max silence at 2 seconds, couldn’t someone leave a 30-second
 message, pause for a couple seconds to gather their thoughts or dig up a
 phone number, and get hung up on?


Two seems short, but nobody has complained about our setting of three.

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Re: [asterisk-users] Capture queue agent drop and put caller back in queue

2013-01-21 Thread Carlos Alvarez
On Mon, Jan 21, 2013 at 11:03 AM, Mitch Claborn mitch...@claborn.netwrote:

 How can I accomplish my goal?


http://lmgtfy.com/?q=battery+backup


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Re: [asterisk-users] recrding calls

2013-01-18 Thread Carlos Alvarez
On Fri, Jan 18, 2013 at 6:25 PM, Joseph syscon...@gmail.com wrote:

 I would like to outgoing/icoming calls and email the files.
 This is what I have:
 ...
 exten = _7.,n,Set(CALLFILENAME=${**EXTEN:1}-${TIMESTAMP})
 exten = _7.,n,Monitor(wav,${**CALLFILENAME},m)
 ...

 How do I email these file?


This is how we do it:

exten =
_1NXXNXX,1,Set(recordfilename=/var/spool/asterisk/monitor/${EXTEN}-${TIMESTAMP:0:8}${TIMESTAMP:8}.WAV)
\exten = _1NXXNXX,n,MixMonitor(${recordfilename},b)
exten = _1NXXNXX,n,(dial here or whatever)
exten = h,1,System(/usr/sbin/sendEmail -t u...@domain.com -f
p...@domain.com-u Call recording for ${recordingfilename} -m There
is a new call
recording. -a ${recordfilename})


Google the sendEmail app and download it, very useful for a lot of things.

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Re: [asterisk-users] recrding calls

2013-01-18 Thread Carlos Alvarez
Yeah, sorry, that example was from one of our older servers.


On Fri, Jan 18, 2013 at 8:08 PM, Warren Selby wcse...@selbytech.com wrote:

 Around version 1.4 or 1.6, TIMESTAMP was phased out and replaced with
 STRFTIME.  See this page for details on how to properly generate a
 timestamp:

 http://www.voip-info.org/wiki/view/Asterisk+func+strftime




 On Fri, Jan 18, 2013 at 8:46 PM, Joseph syscon...@gmail.com wrote:

 On 01/18/13 19:27, Carlos Alvarez wrote:

On Fri, Jan 18, 2013 at 6:25 PM, Joseph [1]syscon...@gmail.com
 wrote:

 I would like to outgoing/icoming calls and email the files.
 This is what I have:
 ...
 exten = _7.,n,Set(CALLFILENAME=${**EXTEN:1}-${TIMESTAMP})
 exten = _7.,n,Monitor(wav,${**CALLFILENAME},m)
 ...
 How do I email these file?

   This is how we do it:
   exten =
   _1NXXNXX,1,Set(**recordfilename=/var/spool/**
 asterisk/monitor/${EXTEN}-
   ${TIMESTAMP:0:8}${TIMESTAMP:8}**.WAV)
   \exten = _1NXXNXX,n,MixMonitor(${**recordfilename},b)
   exten = _1NXXNXX,n,(dial here or whatever)


 Thanks Carlos
 I'm just concentrating right now on ${TIMESTAMP} variable but is is not
 working:

 I have:
 exten = 11,n,Set(recordfilename=/var/**spool/asterisk/monitor/${**
 EXTEN}-${TIMESTAMP:0:8}${**TIMESTAMP:8}.WAV)
 exten = 11,n,MixMonitor(${**recordfilename},b)

 and the file name I got was: -11.wav

 Why I'm not getting any timestamp?


 --
 Joseph

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 Thanks,
 --Warren Selby, dCAP
 http://www.SelbyTech.com http://www.selbytech.com

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[asterisk-users] Call parking in a multi-tenant system

2013-01-15 Thread Carlos Alvarez
We use Asterisk as a hosted PBX.  We've had a couple of requests for
parking, but none of the documentation shows any way to make it aware of
contexts or otherwise make it multi-tenant.  Have I missed something and
does anyone know how to make this work?  Would be on Asterisk 1.6 for now,
1.8 some time soon.

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Re: [asterisk-users] block one number in incoming calls

2013-01-14 Thread Carlos Alvarez
So I'm not the only one who uses the monkeys as our place to send bad calls to.


-- 
Sent from my iPhone

On Jan 14, 2013, at 10:02 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote:

 On Monday 14 January 2013, Salaheddine Elharit wrote:
 i think i didn’t explain correctly may question

 i revive a lot of calls from this number _0666XX and i wants to block
 it to call my number 520xx .

 Use something like
 Exten = _520X./0666XX,1,Answer()
 Exten = _520X./0666XX,n,PlayBack(tt-monkeys)
 Exten = _520X./0666XX,n,HangUp()

 Now when a call comes in from 0666XX to _520X. they will get monkey
 noises.


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Re: [asterisk-users] block one number in incoming calls

2013-01-14 Thread Carlos Alvarez
On Mon, Jan 14, 2013 at 10:48 AM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:

 On Monday 14 January 2013, Carlos Alvarez wrote:
  So I'm not the only one who uses the monkeys as our place to send bad
 calls
  to.

 I actually thought of using a sample from a well-known song by Australian
 singer Kevin Bloody Wilson, but decided that might be a bit *too*
 offensive.
 Besides which, the monkeys sample was already there .


For the really obnoxious problem callers, I typically find out their
corporate phone number and just forward all the marketing calls back to
them system-wide.  I did that with the car warranty people.  I wish I could
have legally recorded them, that could have been fun stuff.

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Re: [asterisk-users] How often to restart Asterisk...

2013-01-11 Thread Carlos Alvarez
On Fri, Jan 11, 2013 at 2:06 PM, Kevin Larsen 
kevin.lar...@pioneerballoon.com wrote:

 However, this does make me wonder, do you restart periodically to try to
 avoid issues or do you just let things run until there is a problem? This
 box had 119 days of up time on the Asterisk process. I have a client that I
 installed an Elastix instance on and the last time I checked it, it was up
 to almost 500 days of up time without an asterisk restart.


I've had boxes run for years, and others have problems in a month or two.
 I have a general practice of having a reboot cron job on critical servers
at 3am on Sunday.  Our customer SLA allows for a maintenance period during
this time.

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Re: [asterisk-users] DIDForSale spam

2013-01-10 Thread Carlos Alvarez
On Thu, Jan 10, 2013 at 10:18 AM, Matthew J. Roth mr...@imminc.com wrote:


 You already have all of our addresses.  Please unsubscribe us.


It's the only way to partially redeem yourself on this list.

Do you really want to be pissing off some of the most active Asterisk
users?  You've already guaranteed that a lot of us will actively tell
people not to do business with you.  I've had the opportunity to warn two
potential users in the last year.

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Re: [asterisk-users] DIDForSale spam

2013-01-10 Thread Carlos Alvarez
On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire r...@coopvr.com wrote:

  It really didn't bother me as much as reading all the posts but that's
 just me...now back to Asterisk issues :)


Sorry to add another, but for me, the main point is that this activity
speaks to the character, ethics, and trustworthiness of the company doing
it.  We all have spam filters.  I just also add the company to my do not
buy/do not recommend list.


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Re: [asterisk-users] DIDForSale spam

2013-01-10 Thread Carlos Alvarez
Hopefully it's not, What is the best DID provider for Asterisk...


On Thu, Jan 10, 2013 at 1:37 PM, Steve Totaro 
stot...@totarotechnologies.com wrote:

 So what asterisk issue do you have?  Let's fix it.

 On Thu, Jan 10, 2013 at 1:49 PM, Ron Wheeler
 rwhee...@artifact-software.com wrote:
  That does not solve any asterisk issue that I have.
 
 
 
  On 10/01/2013 1:32 PM, Carlos Alvarez wrote:
 
  On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire r...@coopvr.com wrote:
 
  It really didn't bother me as much as reading all the posts but that's
  just me...now back to Asterisk issues :)
 
 
  Sorry to add another, but for me, the main point is that this activity
  speaks to the character, ethics, and trustworthiness of the company doing
  it.  We all have spam filters.  I just also add the company to my do not
  buy/do not recommend list.
 
 
  --
  Carlos Alvarez
  TelEvolve
  602-889-3003
 
 
 
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  --
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  President
  Artifact Software Inc
  email: rwhee...@artifact-software.com
  skype: ronaldmwheeler
  phone: 866-970-2435, ext 102
 
 
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Re: [asterisk-users] DIDForSale spam

2013-01-10 Thread Carlos Alvarez
On Thu, Jan 10, 2013 at 3:09 PM, C. Savinovich
c.savinov...@itntelecom.comwrote:


 Although people complaining of spam may be valid, the one part of
 complaining about spam that bothers me, is that some people should look at
 themselves in the mirror and ask the question aloud Why does it bothers me
 to see another Asterisk professional compete with me for jobs?, Why do I
 get angry at seeing someone else's solicitation for services I too provide?

 In reality, anyone who solicits customers in this list does a folly thing
 because this list has more asterisk consultants than customers.   This list
 is the equivalent of a Home Depot parking lot full of construction workers
 looking for a job.


A lot of people on this list are potential customers of the company in
question.  I doubt that most of us compete with them.  We used to be their
customer.


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[asterisk-users] Playing music through VoIP handsets while on hook

2013-01-10 Thread Carlos Alvarez
This is something I've seen with some key systems and PBXs.  When the
phones are on-hook, they can play music throughout the office instead of
having an overhead speaker system do it.  Never heard of it being done with
VoIP, but figured I'd ask if anyone else has.  I don't see any way to do
this on any phones I've looked at.


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Re: [asterisk-users] Playing music through VoIP handsets while on hook

2013-01-10 Thread Carlos Alvarez
On Thu, Jan 10, 2013 at 6:42 PM, Christopher Harrington ch...@acsdi.comwrote:


 Wow, that seems wildly bandwidth inefficient. Is it possible to do
 multicast VoIP?


Depends on whether the phones are local to the server.  Unless you're
looking at hundreds of phones, a 100MB network running 80k to every phone
wouldn't even notice.

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Re: [asterisk-users] DIDForSale spam

2013-01-09 Thread Carlos Alvarez
I don't buy from spammers. Buying email lists may be working for you, but
some people will never do business with you because of it. We cancelled our
account with you because even as a customer we were subjecting to your
unscrupulous marketing.


-- 
Sent from my iPad

On Jan 9, 2013, at 7:10 PM, Jai Rangi jpra...@didforsale.com wrote:

Don,
I have removed yours right away.

Yes, I agree, But just like any company we have purchased/collected email
from different source. Also just like any company we are not perfect, we
make mistakes.

-Jai Rangi


On Wed, Jan 9, 2013 at 5:57 PM, Don Kelly d...@donkelly.biz wrote:

 Jai,

 ** **

 It should not be necessary for me to remove my email address from your
 list. It should not be on there to start with—we do not have, and have
 never had, a relationship that justified you sending me email.

 --Don

 Don Kelly

 PCF Corp
 People Come First
 651 842-1000
 651 842-1001 fax

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jai Rangi
 *Sent:* Wednesday, January 09, 2013 7:50 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] DIDForSale spam

 ** **

 Guys,
 Since I am attached to did for sale:
 My apology to every one who received the DIDForSale 2012 Achievement
 email and you hated it.

 As a asterisk user my question will be.
 If some xyz company send you a so called spam email, what made you think
 that you should spam the mailing lists. I am sure we all get lots if spam
 emails every day. If you really got some time and talent, why don't you
 write some good tips and tricks about asterisk.

 Long story short We have a link where you can unsubscribe your email for
 any further communication.
 http://www.didforsale.com/unsubscribe.php  or Send me your email  address
 I will personally take care of that and will remove your email. This will
 take less than 5 seconds.
 I am sure there will be lot of arguments on why you should that and all. I
 will refrain myself on any further unproductive communication.

 Happy new year to you all.


 

 On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani mi...@enterux.in wrote:***
 *

 +1 here.

 On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com
 wrote:

 On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote:
  On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote:
  What were the senders IP(s)?
 
  Will have to look it up when I get home.
 
  Doug
 
  --
  Ben Franklin quote:
 
  Those who would give up Essential Liberty to purchase a little
 Temporary Safety, deserve neither Liberty nor Safety.
 
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  I have gotten hit with this twice so far. in March and Today:
 
  Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com
  3/8/12
 
  DIDForSale donotre...@didforsale.com via mail.bingotelecom.com
  1/9/13
 
  UGH, when I asked in March where he got my email he said:
 
  Hi Chris,
  We got your contact from the Internet. Let me know the good time to
  talk about this in detail.
  Thank you,
  -Rohit Dhaka
 

 Obviously by harvesting these lists.  I received 2 myself.

 Thanks,
 Steve T

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Re: [asterisk-users] Paging unit suggestions

2013-01-07 Thread Carlos Alvarez
On Mon, Jan 7, 2013 at 1:10 PM, Doug Lytle supp...@drdos.info wrote:

 We currently have an Asterisk system that is hooked up to our old paging
 speakers via sound card, plugged into two amps.

 Each amp drives up to 8 analog speakers in each warehouse (we have 2).
 Both warehouses are around 30k square feet.  Both have a large number of
 printing presses.

 The computer system is that is running Asterisk is around 10 years old and
 starting to fail.  I'm looking to replace both the system and hopefully
 move to a IP paging system, but wanted to reuse the current speakers.


If the amps are good, you could just drive them from a cheap phone with a
regular headset jack.  Set it to auto-answer, send the paging extension to
it.


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Re: [asterisk-users] Verizon SIP trunking Field Trial

2013-01-04 Thread Carlos Alvarez
On Fri, Jan 4, 2013 at 11:18 AM, Eric Wieling ewiel...@nyigc.com wrote:

 Trust me, Verizon doesn't really provide support.What they will do is
 tell you something different (often conflicting stuff) when you send in a
 ticket.One time they tell us the From must be in e.164 format, other
 times they say it does not.We asked for an updated Interop guide weeks
 ago and they have not provided us anything.  We have been with VZ SIP for
 years so I wanted an updated interop guide so we can point them to it when
 they tell us something which conflicts with their docs.  Don't get me
 started on trying to upgrade our service with them..


Sounds like the same huge effort it takes to work with Qwest/Centurylink,
and in the long run we found it simply isn't worth it.  The few benefits of
working with an RBOC are countered by the many drawbacks of working with an
RBOC.

Also we recently acquired a half million minutes/mo from a company who was
tired of dealing with Qwest SIP.  They said the same thing I said above.

I suppose the point of what I'm saying is you should really think about
what you think you will gain from a relationship with them, and whether all
this is worth it (all this means now and how their attitude will affect
you forever).


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Re: [asterisk-users] Verizon SIP trunking Field Trial

2013-01-03 Thread Carlos Alvarez
On Thu, Jan 3, 2013 at 8:13 AM, Michael L. Young myo...@acsacc.com wrote:


 Where I am at is that they want us to use an SBC.  One engineer asked
 about Cisco Call Manager.  I told them that basically if I can accomplish
 the same thing with a Linux box (routing box and sip proxy box) without
 having to spend money on SBCs or expensive Cisco gear, that is the route we
 would like to go.  We are looking at the possibility of handling 140
 concurrent calls... that is what they are designing on their end as well.

 So, I am asking the community for any input.  I have read on here and seen
 on IRC that some in the community are successfully using Asterisk with
 Verizon SIP.  Verizon was going to check and see if they have any notes
 about that and those particular setups.  Can anyone help share any
 information or tidbits on how they were able to sucessfully work with
 Verizon?


It may be too late for this, but in working with another RBOC who didn't
want to deal with Asterisk, I just asked what they do support, and modified
the headers sent by Asterisk to claim that it was one of the devices on
that list.  Done.

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Re: [asterisk-users] Top Posting

2013-01-02 Thread Carlos Alvarez
On Wed, Jan 2, 2013 at 11:02 AM, Ira i...@extrasensory.com wrote:


 And I started communicating with a 2400 baud modem so trimming was a
 necessity and a requirement of friendship.


Bah, spoiled kids.  Mine was a 110 baud acoustic.


 I think the Will Asterisk run on a Rasberry Pi thread the perfect
 example of why this list is dying.


The number of questions posted here that are easily answered with a search
or which are far too basic and open (how do I make Asterisk work) is very
high these days, and that does kill a list.  A lot of us are interested in
helping people who help themselves, and solving complex problems.  I've
seen many tech lists die off when people stop trying to help themselves and
ask intelligent questions.

As to top-posting, another example of when I think it's generally
acceptable is people using tablets.  I have found no way on either my iOS
or Android tablets to quickly/easily post in the traditional manner.  If
I'm faced with spending a few minutes carefully trimming a useful reply or
just not posting it at all, I'm likely to choose the latter if I'm on a
list that says absolutely never top post.

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Re: [asterisk-users] Auto ban IP addresses

2013-01-02 Thread Carlos Alvarez
On Wed, Jan 2, 2013 at 3:49 PM, Frank fr...@efirehouse.com wrote:

 Greetings all,

 I have been seeing a lot of

 [Jan  2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite:
 Sending fake auth rejection for device 100sip:100@108.161.145.18;**
 tag=2e921697

 in my logs lately. Is there a way to automatically ban IP address from
 attackers within asterisk ?


http://www.fail2ban.org/wiki/index.php/Asterisk


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Re: [asterisk-users] Top Posting

2012-12-30 Thread Carlos Alvarez
There's nothing wrong with the list.  The current whining will die off and
things will be back to normal shortly.  Meanwhile I was tempted to bin the
entire thread then realized there's some funny human psychology to be
laughed at.

Top posting to make a few people crazy.  Normally I wouldn't.


On Sun, Dec 30, 2012 at 7:37 PM, James Mortensen 
james.morten...@voicecurve.com wrote:

 I have an idea! Instead of arguing over whether or not top posting or
 bottom posting is the way to go, something that obviously no one will *
 ever* agree on, why not move to Google Groups instead (or something
 similar to Google Groups).

 When I post to Doubango's list, it's easy, there's no top or bottom
 posting wars, it just works. In fact, in a thread, Google Groups usually
 drops you right to the most recent message, so the people who like top
 posting can still see the most recent message while the bottom posters will
 still see the bottom posting format.

 It's either this, or we can sit and watch intelligent people continue to
 degrade one another and argue over something with no agreement in site. :)

 When I mentioned this before, someone from Digium said this will never
 happen, and it's unfortunate.  Maybe they just like to see people bicker
 and argue.

 If there's a better alternative to Google Groups, or a way to set
 preferences in the mailing list so that everyone is happy, maybe that's
 something that could be done?

 James

 On Sun, Dec 30, 2012 at 6:30 PM, Ron Wheeler 
 rwhee...@artifact-software.com wrote:



 On 30/12/2012 11:13 AM, Patrick Lists wrote:

 On 12/30/2012 04:26 PM, Ron Wheeler wrote:

 I participate in a lot of lists and top posting is now the norm since
 people want to see quickly if the message is worth reading.


 Isn't it a bit of a stretch to extrapolate your experience with your
 lists to top posting being the norm? I am subscribed to several lists and
 bottom posting, proper trimming and commenting inline is the norm there.

 Actually the norm is determined by the list rules. If the list rules say
 one must use bottom posting then one should use bottom posting. If someone
 does not like that then don't subscribe, find another source to ask a
 question (the forum, LUG, hire a consultant) or just bottom post.

 Questions come before answers.
 Answers come after questions.

 -1 against changing rule #5.

 Regards,
 Patrick

  Not really enough time in the day to keep track of different rules for
 all the forums.
 I am more concerned about content than form.

 As long as the questions get answered, I can figure out where it is but
 it is a PITA to scroll down through an e-mail to find out that there is
 nothing there worth reading.
 I get over 100 e-mails per day that make it through my filters. I like to
 read the content as soon as it pops up rather than searching for the text.

 Ron



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Re: [asterisk-users] Asterisk with Cisco 887M

2012-12-27 Thread Carlos Alvarez
On Thu, Dec 27, 2012 at 9:10 AM, Edwin Quijada
listas_quij...@hotmail.comwrote:

  Hi!
 I am installing asterisk with my ISP but he give me a Cisco 887M router to
 use for SIP conection. My problem is that I dont know how to link Asterisk
 with this device because I dont have user/pass to use.

 Anybody has a cluee to use CISCO 887M with Asterisk ?


What are the symptoms of the problem?  One way audio?  No audio?


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Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Carlos Alvarez
On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar maill...@lightspeed.cawrote:

 This past holiday weekend has resulted in some real groaners when it comes
 to bugs in our dialplan, making obvious the need for some changes in our
 procedures.

 First, our hours of operation for Christmas Eve, Christmas, Boxing Day and
 New Year's Eve had changed with little to no notice. Okay, fine, whatever,
 I fix.


Boxing day???  Seriously?  There's a holiday for people who beat each other
up?  TIL.

But anyway the best way to test time-based rules is on a VM that has a copy
of your configs, and just change the time.  You can easily run a small VM
to accomodate a copy of your main server on almost any computer.

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Re: [asterisk-users] Call Forwarding / Follow-Me on PRI

2012-12-27 Thread Carlos Alvarez
On Thu, Dec 27, 2012 at 2:41 PM, Barry D. Hassler
barry.hass...@gmail.comwrote:

 Friends,

 Curious if others have run into this scenario, and can shed further light
 on it. I am working with an installed base of systems using PRI circuits
 from several carriers, and the symptoms I relate occur across the board.


We have encountered it, and simply told the carriers to stop blocking it or
lose the business.  All but one did it, and we dropped their services.
 Don't know that there's a good work-around otherwise.

Is there a reason you don't just go all SIP, where 98% of the service
providers will accept any CLID?

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[asterisk-users] Change phone display from queue calls

2012-12-06 Thread Carlos Alvarez
We are trying to set up a system where the calls from the queue show a
specific name or number on the phone.  The calls would come into one of a
few dozen DID numbers, each one for a specific company.  The agent needs to
know which company the call is for and answer appropriately.  I've done a
lot of this in dialplans but haven't found a way to do it in a queue.

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Re: [asterisk-users] Need qualifications of SIP trunk providers

2012-11-30 Thread Carlos Alvarez
On Fri, Nov 30, 2012 at 7:10 AM, Daniel - Asterisk earohua...@gmail.comwrote:

 Thank you Carlos,

 What does mean 'por-out'?
 I'm expecting 1 min/month in  out.


Port out means a number was ported to another carrier.

10k minutes is not huge but a decent number that should get you a
reasonable rate with the carrier of your choice.  Vitelity is a good fit
for that size.  I can't say they are better than the others because I
haven't used them, but we had a few hundred DIDs and probably did 50k
minutes with them at one point.  As we've grown we have moved to others
(we're around half a million min/mo now).

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Re: [asterisk-users] Need qualifications of SIP trunk providers

2012-11-29 Thread Carlos Alvarez
On Thu, Nov 29, 2012 at 3:22 PM, Daniel - Asterisk earohua...@gmail.comwrote:

 Hello List,

 Since I'm looking for a new VoIP provider for US origination/termination,
 I will very appreciate if you can chare your experience with Flowroute,
 Vitelity and Voip.ms


Vitelity is reliable and decent, but no phone support.  Have not used the
others.

Oh also if you lose a number on Vitelity to a port-out, they won't know and
won't stop billing you for it.

What's your expected volume in/out?

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Re: [asterisk-users] watchdog like functions

2012-11-20 Thread Carlos Alvarez
Switching to SIP is likely your best solution.  IAX is buggy.  Always has
been, and I'll bet always will be.


On Tue, Nov 20, 2012 at 7:34 PM, asterisk asterisk aster...@ck-lee.comwrote:

 I wish to ask if there is way to keep IAX trunk connection up. I have a
 small server on Xen VPS but notice that my IAX trunk drops after some time.

 I understand there is cron job to function as sip watchdog.

 My asterisk is 11.0.1

 Thanks for suggestions.

 CK

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Re: [asterisk-users] 3G Quality

2012-11-14 Thread Carlos Alvarez
On Wed, Nov 14, 2012 at 3:15 PM, Roy Abshire r...@coopvr.com wrote:

 Has anyone been able to configure Asterisk to work over 3G?

 I bought Nokia Cell Phones just for that purpose and they register fine
 over WiFi and 3G but the quality is just not good enough and sometimes the
 call just disconnects.


VoIP does not work reliably over 3G.  I've had best success with g.729 but
it's still not great.

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Re: [asterisk-users] 3G Quality

2012-11-14 Thread Carlos Alvarez
On Wed, Nov 14, 2012 at 3:30 PM, Roy Abshire r...@coopvr.com wrote:

  Ok, what about 4G, I tried it with my 4G ATT Hotspot too.  I get 4 full
 signal and the call goes through but keeps cutting out or dropping.

 I'm using the ATT Elevate with a Nokia N97.  SIP connects fine, I can
 make and receive calls and when I check the console it connects using ilbc
 but we have a hard time hearing each other clear and it drops unexpectedly
 without hanging up the other side.


I wouldn't expect any cellular connection to pass VoIP reliably.  Think
about it.  The phone call is on the same airwaves but is prioritized.  The
data gets the lowest priority.  Again, all my testing has left me using
only g.729 when I really need to do VoIP over cellular, but it's never been
perfect.

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Re: [asterisk-users] 3G Quality

2012-11-14 Thread Carlos Alvarez
On Wed, Nov 14, 2012 at 3:34 PM, Roy Abshire r...@coopvr.com wrote:

  G729 requires a paid license right?  If I wanted to test it using
 G729...do I still have to buy a license for it?

 My goal is to use my cell phone for Text Only and pay $15/month and use my
 ATT WiFi Hotspot for VOIP calls wherever I go because I rely on my VOIP
 for business.


It's a $20 license, dirt cheap.

If your goal is to simply save the $45/mo it costs for a full unlimited
plan, you're wasting a lot of time and energy.

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Re: [asterisk-users] 3G Quality

2012-11-14 Thread Carlos Alvarez
On Wed, Nov 14, 2012 at 4:20 PM, Roy Abshire r...@coopvr.com wrote:

  My goal is to always be connected to my VOIP system using 3G or 4G.  I'm
 still going to have to pay the ATT Data only $50/month plan for 4G and I
 get 5gb a month which is plenty.


Why would you always want to be connected to the VoIP system?  Just do
concurrent ringing to the cell number like almost everyone does.  Yeah, I
wish the VoIP over cell data fantasy worked also, but it really doesn't, as
far as I've seen and heard.  Particularly while moving.

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Re: [asterisk-users] 3G Quality

2012-11-14 Thread Carlos Alvarez
On Wed, Nov 14, 2012 at 5:22 PM, Roy Abshire r...@coopvr.com wrote:

  Believe me, there is a method to my madness that I didn't want to get
 into but here it goes.


Quite often, telling people the core problem can lead to better solutions.
 I assumed you wanted mobile, but fixed cellular could work for this.


 I want to get this to work reliably even with low quality and bandwidth so
 I can install VOIP Phones at my vacation rental properties that can only
 get 4G or low speed Satellite with only 128k upload.


128 upload is no problem, but latency on geosynchronous satellite is
horrible.  Pretty much not usable for voice.  LEO satellites like Iridium
and Inmarsat work, but are VERY expensive.

Have you checked out the new Dish network satellite internet service?  I
haven't used it, but the specs look great and the price is cheap.


 I went up and stayed at one of the properties and bought my ATT 4G Elevate
 device just to test it out over VOIP at the cabin and I got 2 bars and my
 Nokia connected and registered over SIP...but calling was garbled.


What you need is an external antenna, possibly a directional one if you
know the general direction of the cell tower.  Wilson makes some great
amplifiers and antenna systems.  I use a Wilson amp on my boat with a tall
omni marine antenna, and it lets me make calls over VoIP where there would
otherwise be no signal (they are choppy but usable).

There is some controversy on the best CODEC, but my experience is that
g.729 is best on low-quality connections.

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Re: [asterisk-users] Static on calls - v1.8.15.0

2012-11-08 Thread Carlos Alvarez
On Thu, Nov 8, 2012 at 9:34 AM, James Lamanna jlama...@gmail.com wrote:

 Hi,
 I'm testing out a server with asterisk 1.8.15.0 on it.
 I'm experiencing static occurring on almost 90% of calls on this
 particular server.
 All test phones are using SIP, and calls to/from PSTN servers are
 delivered using IAX2.

 I have other production servers running 1.4.x that do not have this issue
 that use the same PSTN connections.
 I haven't seen any ethernet errors or anything like that. Load is minimal
 since this is still a test server.
 The server itself has 16GB of RAM and Dual Quad Core Xeon E5345s.

 I'm sort of baffled as to where to start looking for the root cause of
 this issue, but it appears to be isolated to only this machine.


I would recommend you try SIP and lock out all CODECs on both sides except
ulaw.  See what happens.

We've had this problem both because of CODED mismatch and just due to IAX's
many many bugs.


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Re: [asterisk-users] forwarding all calls to cells

2012-11-06 Thread Carlos Alvarez
On Tue, Nov 6, 2012 at 5:33 PM, Noam Birnbaum
n...@maccentricsolutions.comwrote:

 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?


What a waste of effort, bandwidth, and money.  There are a dozen services
out there that can do this far more efficiently and for a lot less money.
 You will be using bandwidth in both directions and double the channels
(one call in, forwarded call out).  It will increase the latency by a lot
and the chance of call quality problems is huge.  Call in over a cheap
cable connection, call out on the same, then over a cell.

Concurrent ringing with cells is popular and makes a lot of sense since you
primarily typically answer on the desk phone.  If it's just forwarding,
then just use a forwarding service which will cost less and not introduce
the extra cable network layer.

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Re: [asterisk-users] multitenanat third party app

2012-10-31 Thread Carlos Alvarez
Indeed this is getting ridiculous.  This person also called me (!!) for
some free consulting after I had posted the answer a few days ago.

NOTE:  We aren't going to engineer your system for you!  We as a group will
provide help and some basic code to get you started.  If you don't know how
to start working with the fully working stuff I provided already, you're
not ready to deploy a system this complex.


On Wed, Oct 31, 2012 at 2:59 PM, Mitul Limbani mi...@enterux.in wrote:

 Stop asking same questions !!!
 On Oct 31, 2012 11:54 PM, Darin Iv adari...@gmail.com wrote:

 Is it possible to bul multitenant system using some third party opensouce
 application My design is like this.

 Company A:
 Context Company_A
 IVR Company A
 Extensions: 101,102,103,104 etc.

 Company B:
 Context Company_B
 IVR Company B
 Extensions: 101,102,103,104 etc.

 Company C:
 Context Company_C
 IVR Company
 Extensions: 101,102,103,104 etc.


 Company D:
 Context Company_D
 IVR Company D
 Extensions: 101,102,103,104 etc.

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Re: [asterisk-users] multi tenant

2012-10-30 Thread Carlos Alvarez
On Tue, Oct 30, 2012 at 12:15 AM, Darin Iv adari...@gmail.com wrote:

 Hi all,

 I need to configure DIDs for different companies and they should reach on
 different extension with different context. Cant we have same extension
 in different context?

 This is what we we want
 Company A:
 Context Company_A
 IVR Company A
 Extensions: 101,102,103,104 etc.

 Company B:
 Context Company_B
 IVR Company B
 Extensions: 101,102,103,104 etc.


There are multiple ways to do this.  One way is the Local dial.  We have
done this for companies who are different entities but want to do 3-digit
dial.

Dial(Local/101@company_a#extensions,25)

Where we assume you have a context like:

[company_a#extensions]

exten = 101,1,Dial(SIP/company_a.${EXTEN},25)

Another way is to simply do an include for the other company's extension
context.  However that requires that you not duplicate the extension
numbers between the contexts/companies.


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Re: [asterisk-users] multi tenant

2012-10-30 Thread Carlos Alvarez
I am attempting to send this again.  The mail processor is interpreting the
Asterisk commands in my message as mail processor command and bouncing the
message.  That's why where is junk before many of the lines below.

On Tue, Oct 30, 2012 at 12:15 AM, Darin Iv adari...@gmail.com wrote:

 Hi all,

 I need to configure DIDs for different companies and they should reach on
 different extension with different context. Cant we have same extension
 in different context?

 This is what we we want
 Company A:
 Context Company_A
 IVR Company A
 Extensions: 101,102,103,104 etc.

 Company B:
 Context Company_B
 IVR Company B
 Extensions: 101,102,103,104 etc.


There are multiple ways to do this.  One way is the Local dial.  We have
done this for companies who are different entities but want to do 3-digit
dial.

...  Dial(Local/101@company_a#extensions,25)

Where we assume you have a context like:

...  [company_a#extensions]

...  exten = 101,1,Dial(SIP/company_a.${EXTEN},25)

Another way is to simply do an include for the other company's extension
context.  However that requires that you not duplicate the extension
numbers between the contexts/companies.
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Re: [asterisk-users] high capacity analog - sip gateway

2012-10-25 Thread Carlos Alvarez
On Thu, Oct 25, 2012 at 1:21 PM, Justin Killen 
jkil...@allamericanasphalt.com wrote:

  I’m looking for an fxs - sip gateway/router/switch for about 100
 existing analog phones.  I’d like to get this done cheaply, but I want to
 make sure that whatever we buy works well with asterisk as well.  As far as
 I can tell, digium make no such device.  The only ones I’ve been able to
 find with a 48 port capacity are these two:


I have a deployment of 96 analog ports using a Digium T1 card ($500 on
eBay) and Rhino analog channel banks (also cheap on eBay).  We have
extremely high reliability from this configuration.  In fact, other than
the normal analog annoyances like occasional echo, they are rock solid.

Are you doing this instead of VoIP phones for cost reasons?

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Re: [asterisk-users] high capacity analog - sip gateway

2012-10-25 Thread Carlos Alvarez
On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen 
jkil...@allamericanasphalt.com wrote:

 **

 Cost and ease of deployment, yes.  At this specifc location we are
 currently using Centrex lines (ATT hosted) and are looking for a way to
 move into something cheaper without throwing away the existing phones.  I
 like the idea of using a channel bank – I’ll look into that as an option as
 well.


You should be able to also connect the Centrex lines to the channel banks,
I believe.

I always advocate throwing out old analog phones as they will be a pain,
but understand if you absolutely cannot.  Just keep in mind you can get a
decent VoIP phone for $60 that is very likely to be nicer than what they
have now and do much more.

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Re: [asterisk-users] high capacity analog - sip gateway

2012-10-25 Thread Carlos Alvarez
On Thu, Oct 25, 2012 at 2:14 PM, Christopher Harrington ch...@acsdi.comwrote:

 On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez car...@televolve.comwrote:

 I always advocate throwing out old analog phones as they will be a pain,
 but understand if you absolutely cannot.  Just keep in mind you can get a
 decent VoIP phone for $60 that is very likely to be nicer than what they
 have now and do much more.


 Out of curiosity, would you mind sharing that with us?


The phone?  Grandstream.  They have people who love them and hate them, but
so far we're pretty happy with them.  A GXP2124 is on my desk right now,
and I have access to any phone I want.  I like it.  There are low-end
models still with a display for under $60.


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Re: [asterisk-users] high capacity analog - sip gateway

2012-10-25 Thread Carlos Alvarez
On Thu, Oct 25, 2012 at 2:34 PM, Justin Killen 
jkil...@allamericanasphalt.com wrote:

 ** ** **

 I think if we were to go to VoIP phones, one thing that we would have to
 consider very highly in a phone would be that they have VLAN settings and a
 built-in Ethernet hub/switch so that we can just inject it into the user’s
 computer LAN connection.  The cost and time of rewiring some of these
 locations is not something we’re comfortable with doing.


The Grandstreams do that.  Most phones do.


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[asterisk-users] Voicemail to text for Asterisk

2012-10-22 Thread Carlos Alvarez
A customer has asked us to provide that feature.  I know there are a few
methods and products out there, but I haven't paid attention in a while.
 It is for about 300 users, and we'll consider open as well as paid-for
products.  We would prefer to pay for supported products as the cost will
be passed on to the customer and they are willing to pay for quality.  Do
not want any complex scripting screwing around with third parties and such.
 Your ideas welcome.

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Re: [asterisk-users] Voicemail to text for Asterisk

2012-10-22 Thread Carlos Alvarez
On Mon, Oct 22, 2012 at 12:40 PM, Christopher Harrington ch...@acsdi.comwrote:

 All automated solutions -- paid or free -- are terrible. The technology
 simply does not exist at this point at a level that is acceptable to most
 customers. If quality is paramount, you are better off doing the
 transcription in-house with a human.


In-house transcriptions are definitely out of the question, but any
experience with outsourced solutions would be useful.  As far as I can tell
the current service is automated, and as awful as Google Voice, yet they
find it useful.  Their existing carrier uses Broadsoft and I'm not sure if
they have that built in.

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Re: [asterisk-users] Voicemail to text for Asterisk

2012-10-22 Thread Carlos Alvarez
On Mon, Oct 22, 2012 at 2:43 PM, Nickolay V. Shmyrev
nshmy...@nexiwave.comwrote:

 There is no holy grail yet, speech technology deployment requires a
 close cooperation between the speech technology provider and the users.
 It's not plug and play but after some joint efforts automated
 transcriptions must be useful.

 If anyone wants to experiment with CMUSphinx-based automated solution to
 transcribe voicemails, drop me a note. The results could be pretty
 interesting.


We will probably give it a try, though not sure when.  Probably in about a
month.


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Re: [asterisk-users] Not able to post to list

2012-10-09 Thread Carlos Alvarez
On Tue, Oct 9, 2012 at 12:16 PM, Adnan 112linuxstockh...@gmail.com wrote:

 Hi
 who is responsible for this mailing list? i am not able to post to it.


You just did.

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Re: [asterisk-users] Fax for Asterisk success rates?

2012-10-04 Thread Carlos Alvarez
On Thu, Oct 4, 2012 at 6:29 AM, Brett Lehrer brett.leh...@solarismed.comwrote:

 I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking
 service over a DSL line solely dedicated to VoIP usage.  For both incoming
 and outgoing faxes, I'm getting a failure rate of just over 25%, and over a
 handful of reasons.

 Is it natural to have this many problems on a completely digital
 configuration?  I'm trying to cut our analog phone line (because it's so
 expensive), but some fax machines just don't seem to ever accept a fax.
  Many of the failures are on the same numbers, forcing me to fall back to
 an old analog fax machine just to make sure it actually gets through.

 Has anyone else had any similar experiences, or is this indicative of a
 failure in the setup on my end (or even the trunking service)?


I'm not going to address the tech issues, as others already have.  And if
you didn't know, Steve Underwood is THE fax guy so whatever he says is
gold, listen to him.

However I'd just suggest that you look at the business case for screwing
around with fax at all.  As a society, if we had decided to stop supporting
this dead technology years ago, with all the time and money we've
collectively wasted we could have completely eliminated world hunger.  I
can't count the hundreds of hours I've wasted on fax support just to prop
up this stupid and unnecessary technology.  We just made the decision this
week to outsource it all and never deal with it on our network again.  I am
slowly re-gaining my sanity because of that decision.

Now I'm going to take a fax machine out to the parking lot and shoot it,
even talking about this awful waste of time makes my blood boil.

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Re: [asterisk-users] Fax for Asterisk success rates?

2012-10-04 Thread Carlos Alvarez
On Thu, Oct 4, 2012 at 10:06 AM, Lee Howard fax...@howardsilvan.com wrote:

 I recognize that you're being a bit facetious in this latter comment


No, not really.  I stand by it.  Useless and *should* be dead.  It's dead
and people just don't know it.


 There is no adequate replacement for fax.  E-mail doesn't do it


Yes, it does.



 Well, if you were using stand-alone fax machines then that was part of
 your problem.


That was actually the only part of my post that was in jest.



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Re: [asterisk-users] Call Termination Provider Madness

2012-10-03 Thread Carlos Alvarez
On Wed, Oct 3, 2012 at 7:35 AM, Chris Nighswonger 
cnighswon...@foundations.edu wrote:


 At this point I only have ~40 extensions, so I took Michel's advise
 and set my RTP range to 1-10100. The default 1 ports was a bit
 more surface area than I want to expose.


If you think 100 or 10k RTP ports going to your voice server makes ANY
difference in security, you really need to re-think this and study more.
Not to be a dick or anything, but really, think about it.



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Re: [asterisk-users] Call Termination Provider Madness

2012-10-03 Thread Carlos Alvarez
On Wed, Oct 3, 2012 at 8:38 AM, Chris Nighswonger 
cnighswon...@foundations.edu wrote:

 I'm speaking of surface area. Ask any general if he would rather have
 to defend a 1000 mile front or a 1 mile front. You are right that an
 open port is an open port, but trying keeping the crowd out of 1
 doors is *much* harder than trying to keep them out of 100 doors.


Your trite comparison is irrelevant in this context.  You are not
protecting your 100 ports any more or less than 1000 or 10,000.  But do
as you will, I'll agree to disagree.

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Re: [asterisk-users] Call Termination Provider Madness

2012-10-03 Thread Carlos Alvarez
On Wed, Oct 3, 2012 at 8:46 AM, Eric Wieling ewiel...@nyigc.com wrote:

 A port is not a door if there is nothing listening on the port.

 Open ports are not a security issue.  Stuff running on open ports are.


In other words, a million ports with nothing listening is no worse than one
with nothing listening.


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