Re: [asterisk-users] looking for a way to do appointment reminders

[Yves A.]

Hi Brandon,

as you are asking for professional help for a commercial project, I 
would recommend you to place a bounty.
You can contact me directly if you want my professional help... I have 
developed exactly what you´re looking
for and this solution is running in a high-call-volume installation 
without any issues.

regards,
yves

Am 26.04.2013 03:55, schrieb Brandon Coale:

Hello,

My health care organization is looking for a way to do appointment 
reminders.  We currently have staff members who spend part of each day 
manually calling patients to remind them of their upcoming 
appointments, and we would like to automate this process.


Our electronic health record software would provide such information 
as the patient's name, phone number, and day and time of the 
appointment, and Asterisk could take this information and place an 
automated call to the patient.  We would like the reminder call to use 
text-to-speech to personalize the call, such as "We have an 
appointment reminder for [first name].  The appointment is on [date] 
at [time].


I am wondering if anyone has experience with using Asterisk for this 
type of application, and would be willing to share any details of how 
you implemented it?  I am interested in any ideas, from very simple to 
feature-rich.  We would be doing a new installation of Asterisk for 
this purpose, so we could use any version of Asterisk you would 
recommend.


Thank you!
Brandon


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[asterisk-users] looking for a way to do appointment reminders

[Brandon Coale]

Hello,

My health care organization is looking for a way to do appointment 
reminders.  We currently have staff members who spend part of each day 
manually calling patients to remind them of their upcoming appointments, 
and we would like to automate this process.


Our electronic health record software would provide such information as 
the patient's name, phone number, and day and time of the appointment, 
and Asterisk could take this information and place an automated call to 
the patient.  We would like the reminder call to use text-to-speech to 
personalize the call, such as "We have an appointment reminder for 
[first name].  The appointment is on [date] at [time].


I am wondering if anyone has experience with using Asterisk for this 
type of application, and would be willing to share any details of how 
you implemented it?  I am interested in any ideas, from very simple to 
feature-rich.  We would be doing a new installation of Asterisk for this 
purpose, so we could use any version of Asterisk you would recommend.


Thank you!
Brandon


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[asterisk-users] Users appending # sign when dialing an extension from automated greeting

[Vernon Polinkichov]

We have an automated greeting on our Asterisk phone system, that like many,
has the phrase "if you know your party's extension, you may dial it at any
time".

Today, I encountered a user dialing in from the outside, attempting to dial
an extension, but he was appending the # key at the end.

This, as I would have expected, made Asterisk think he was trying to call
the extension 2134#, which went nowhere.

Personally, unless instructed to do so, typically with the phrase "followed
by the pound sign", I don't enter a pound sign when dialing a user
extension or inputting digits on phone system menus.

Is stripping the # sign off of extensions dialed from an automated greeting
something most phone systems do?

Thanks,

Vernon
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Re: [asterisk-users] Asterisk Calendar integration suggestions

[Hans Witvliet]

Might have a look at tine:
http://www.tine20.org/wiki/index.php/Admins/Asterisk_integration

hw

-Original Message-
From: Steve Totaro 
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Asterisk Calendar integration suggestions
Date: Thu, 25 Apr 2013 10:43:52 -0400

Without knowing requirements, Sugar CRM seems to be the most supported.
 


Thanks,
Steve Totaro


On Thu, Apr 25, 2013 at 9:22 AM, j...@millican.us 
wrote:
Hello all,
I am looking into building a calendar server (due to business
requierments I can not use public hosted calender like Google),
and am looking for suggestions based on experience with
different calendar applications/servers available for Linux that
you have integrated with Asterisk.  If you can give a quick,
simple list of what worked and what didn't I would be very
grateful.
Thank You,
John


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Re: [asterisk-users] h323-sip: one way connection

[Asghar Mohammad]

nuFone h323 or openh323?


On Thu, Apr 25, 2013 at 9:33 PM, s m  wrote:

> flavor? i do not understand what you mean. please explain more.
> thanks
>
>
> On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad wrote:
>
>> what flavor of h323 you are using?
>>
>>
>> On Wed, Apr 24, 2013 at 8:50 AM, s m  wrote:
>>
>>> thanks Asghar,
>>> i do it, but no thing happened:(
>>> asterisk do not identify host line as ip address of the other end
>>>
>>>
>>> On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad wrote:
>>>
 try type=peer instead of friend.


 On Tue, Apr 23, 2013 at 10:04 AM, s m  wrote:

> i know what is the exactly problem. i enable debug for h323 and it
> says:
> "could not find user by name 200 or address 192.168.0.146"
>
> when i change "peer-146" to "200" every thing is ok and i can call
> from two side. but it is not good for me because 200 is the name of
> extension and when i config asterisk systems, i don't know the name of
> extensions, therefore i should use addresses not name of extensions.
> do you know how i should define address of the other end in h323.conf
> file? i define the address by "host=192.168.0.146" but asterisk can not
> find it? why?
>
>
> On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad 
> wrote:
>
>> please post cli output for both calls.
>>
>>
>> On Mon, Apr 22, 2013 at 11:32 AM, s m  wrote:
>>
>>> hello everybody
>>>
>>> i want to have sip connection between two asterisk systems (145 and
>>> 146). connection from 145 to 146 is ok but i can not call from 146 to
>>> 145.
>>> this is h323.conf file in 145:
>>> [peer146]
>>> host=192.168.0.146
>>> type=friend
>>> context=from-trunk
>>>
>>>
>>> [to-146]
>>> type=peer
>>> host=192.168.0.146
>>> faststart=yes
>>> tunneling=no
>>> progress_audio=yes
>>> disallow=all
>>> allow=alaw
>>> allow=ulaw
>>>
>>> this is mu extensions.conf file in 145:
>>>
>>> [from-trunk]
>>> exten=>_1.,1,Dial(SIP/to-231/1${EXTEN:1})
>>> [line-231]
>>> exten=>_2.,1,Dial(H323/to-146/2${EXTEN:1})
>>>
>>> i have this error: dropping call because extensions '100', 's' and
>>> 'i'
>>> doesn't exists in context default".
>>>
>>> if i change "peer146" to "general", every thing is ok and i can call
>>> from two side. my question is: in h323 connection, is it a MUST to
>>> have "general" context in h323.conf? if not, why i have this error
>>> and
>>> how i can solve it?
>>> thanks in advance
>>> sam
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com--
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>>>http://www.asterisk.org/hello
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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>


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>>>
>>>
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>>>
>>
>>
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Re: [asterisk-users] h323-sip: one way connection

[s m]

flavor? i do not understand what you mean. please explain more.
thanks


On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad wrote:

> what flavor of h323 you are using?
>
>
> On Wed, Apr 24, 2013 at 8:50 AM, s m  wrote:
>
>> thanks Asghar,
>> i do it, but no thing happened:(
>> asterisk do not identify host line as ip address of the other end
>>
>>
>> On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad wrote:
>>
>>> try type=peer instead of friend.
>>>
>>>
>>> On Tue, Apr 23, 2013 at 10:04 AM, s m  wrote:
>>>
 i know what is the exactly problem. i enable debug for h323 and it
 says:
 "could not find user by name 200 or address 192.168.0.146"

 when i change "peer-146" to "200" every thing is ok and i can call from
 two side. but it is not good for me because 200 is the name of extension
 and when i config asterisk systems, i don't know the name of extensions,
 therefore i should use addresses not name of extensions.
 do you know how i should define address of the other end in h323.conf
 file? i define the address by "host=192.168.0.146" but asterisk can not
 find it? why?


 On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad 
 wrote:

> please post cli output for both calls.
>
>
> On Mon, Apr 22, 2013 at 11:32 AM, s m  wrote:
>
>> hello everybody
>>
>> i want to have sip connection between two asterisk systems (145 and
>> 146). connection from 145 to 146 is ok but i can not call from 146 to
>> 145.
>> this is h323.conf file in 145:
>> [peer146]
>> host=192.168.0.146
>> type=friend
>> context=from-trunk
>>
>>
>> [to-146]
>> type=peer
>> host=192.168.0.146
>> faststart=yes
>> tunneling=no
>> progress_audio=yes
>> disallow=all
>> allow=alaw
>> allow=ulaw
>>
>> this is mu extensions.conf file in 145:
>>
>> [from-trunk]
>> exten=>_1.,1,Dial(SIP/to-231/1${EXTEN:1})
>> [line-231]
>> exten=>_2.,1,Dial(H323/to-146/2${EXTEN:1})
>>
>> i have this error: dropping call because extensions '100', 's' and 'i'
>> doesn't exists in context default".
>>
>> if i change "peer146" to "general", every thing is ok and i can call
>> from two side. my question is: in h323 connection, is it a MUST to
>> have "general" context in h323.conf? if not, why i have this error and
>> how i can solve it?
>> thanks in advance
>> sam
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
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>
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>


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>>>
>>>
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>>>
>>
>>
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>
>
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>http://www.asterisk.org/hello
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> To UNS

Re: [asterisk-users] Load Balancing

[acheraime]

You the couple opensips + asterisk will help you. Opensips loadbalance module 
is your friend.



Sent from my iPhone

On Apr 25, 2013, at 11:44 AM, Olivier  wrote:

> Hello,
> 
> I've been given the task to study what would a good way to load balance SIP 
> trafic.
> 
> The prospective setup is :
> - call centers sending outbound SIP trafic (no inbound) from SIP devices 
> (with public fixed IP address),
> - a couple of outbound SIP trunks to which trafic from call centers is to be 
> forwarded
> - a load balancing system between call centers and SIP trunks.
> 
> Load balancing system main task is:
> - provide some LCR routing,
> - improve availability.
> 
> Can (should) it be done with Asterisk alone or should I look for other 
> components ?
> 
> Regards
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] Sip and the media path

[Kevin Larsen]

David,

you obviously have to test for your situation, but the short answer is 
that it should. The connection will start with running through Asterisk, 
but very quickly the phones will see that they can talk directly and take 
the Asterisk server out of the media path. There are a couple of gotchas 
that can happen based on your dial options, so check out this page:
http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite 

canreinvite was renamed to directmedia in Asterisk 1.6.2, but the page is 
still pretty good with regards to the options that are available.

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



From:   David Wessell 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
, 
Date:   04/25/2013 10:49 AM
Subject:Re: [asterisk-users] Sip and the media path
Sent by:asterisk-users-boun...@lists.digium.com



Kevin,

Thanks for the info. Clarification. The asterisk server is NOT on the same 
LAN as the phones. The asterisk server is in a datacenter only accessible 
via WAN.

However, all of the phones are in side of the same LAN. Will directmedia 
still function that way?

Thanks
David

From: Kevin Larsen 
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users@lists.digium.com>
Date: Thursday, April 25, 2013 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Sip and the media path

You will want to look at the directmedia option. You will want all the 
phones on the same lan as the Asterisk server to be directmedia=yes and 
the ones on the wan to be directmedia=no. Then, internal calls will send 
the media between themselves without involving Asterisk, but ones outside 
on the wan will be forced to talk directly to the Asterisk server for 
everything. You might also want to look at the nonat option of 
directmedia.

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



From:David Wessell 
To:Asterisk Users Mailing List - Non-Commercial Discussion <
asterisk-users@lists.digium.com>, 
Date:04/25/2013 07:33 AM
Subject:[asterisk-users] Sip and the media path
Sent by:asterisk-users-boun...@lists.digium.com



We're running asterisk 1.8 in the DC on a public IP address.

Connecting to it are about 200 phones behind a LAN in a remote location.

Is there a way to reliably keep asterisk out of the media stream on 
internal calls inside that LAN? All phones are Polycom Soundpoint phones.

Asterisk would say in the media stream for any calls that traverse from 
LAN to WAN. However it would step out for LAN to LAN calls.

Thanks 
David 
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Re: [asterisk-users] Sip and the media path

[David Wessell]

Kevin,

Thanks for the info. Clarification. The asterisk server is NOT on the same LAN 
as the phones. The asterisk server is in a datacenter only accessible via WAN.

However, all of the phones are in side of the same LAN. Will directmedia still 
function that way?

Thanks
David

From: Kevin Larsen 
mailto:kevin.lar...@pioneerballoon.com>>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Date: Thursday, April 25, 2013 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Subject: Re: [asterisk-users] Sip and the media path

You will want to look at the directmedia option. You will want all the phones 
on the same lan as the Asterisk server to be directmedia=yes and the ones on 
the wan to be directmedia=no. Then, internal calls will send the media between 
themselves without involving Asterisk, but ones outside on the wan will be 
forced to talk directly to the Asterisk server for everything. You might also 
want to look at the nonat option of directmedia.

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



From:David Wessell mailto:da...@ringfree.biz>>
To:Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>,
Date:04/25/2013 07:33 AM
Subject:[asterisk-users] Sip and the media path
Sent by:
asterisk-users-boun...@lists.digium.com




We're running asterisk 1.8 in the DC on a public IP address.

Connecting to it are about 200 phones behind a LAN in a remote location.

Is there a way to reliably keep asterisk out of the media stream on internal 
calls inside that LAN? All phones are Polycom Soundpoint phones.

Asterisk would say in the media stream for any calls that traverse from LAN to 
WAN. However it would step out for LAN to LAN calls.

Thanks
David
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[asterisk-users] Load Balancing

[Olivier]

Hello,

I've been given the task to study what would a good way to load balance SIP
trafic.

The prospective setup is :
- call centers sending outbound SIP trafic (no inbound) from SIP devices
(with public fixed IP address),
- a couple of outbound SIP trunks to which trafic from call centers is to
be forwarded
- a load balancing system between call centers and SIP trunks.

Load balancing system main task is:
- provide some LCR routing,
- improve availability.

Can (should) it be done with Asterisk alone or should I look for other
components ?

Regards
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Re: [asterisk-users] Asterisk Calendar integration suggestions

[Steve Totaro]

Without knowing requirements, Sugar CRM seems to be the most supported.

Thanks,
Steve Totaro


On Thu, Apr 25, 2013 at 9:22 AM, j...@millican.us  wrote:

> Hello all,
> I am looking into building a calendar server (due to business requierments
> I can not use public hosted calender like Google), and am looking for
> suggestions based on experience with different calendar
> applications/servers available for Linux that you have integrated with
> Asterisk.  If you can give a quick, simple list of what worked and what
> didn't I would be very grateful.
> Thank You,
> John
>
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
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[asterisk-users] Asterisk Calendar integration suggestions

[j...@millican.us]

Hello all,
I am looking into building a calendar server (due to business 
requierments I can not use public hosted calender like Google), and am 
looking for suggestions based on experience with different calendar 
applications/servers available for Linux that you have integrated with 
Asterisk.  If you can give a quick, simple list of what worked and what 
didn't I would be very grateful.

Thank You,
John


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Re: [asterisk-users] Sip and the media path

[Kevin Larsen]

You will want to look at the directmedia option. You will want all the 
phones on the same lan as the Asterisk server to be directmedia=yes and 
the ones on the wan to be directmedia=no. Then, internal calls will send 
the media between themselves without involving Asterisk, but ones outside 
on the wan will be forced to talk directly to the Asterisk server for 
everything. You might also want to look at the nonat option of 
directmedia.

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



From:   David Wessell 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
, 
Date:   04/25/2013 07:33 AM
Subject:[asterisk-users] Sip and the media path
Sent by:asterisk-users-boun...@lists.digium.com



We're running asterisk 1.8 in the DC on a public IP address.

Connecting to it are about 200 phones behind a LAN in a remote location.

Is there a way to reliably keep asterisk out of the media stream on 
internal calls inside that LAN? All phones are Polycom Soundpoint phones.

Asterisk would say in the media stream for any calls that traverse from 
LAN to WAN. However it would step out for LAN to LAN calls.

Thanks
David
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Re: [asterisk-users] Asterisk 1.8 and 11

[qasimak...@gmail.com]

Read up on new features and changelog of asterisk 11 you'll find the
changes there.

Regards,
Qasim


On Thu, Apr 25, 2013 at 3:32 PM, bilal ghayyad  wrote:

> Hello;
>
> How I can compare between Asterisk 1.8 and 11 with reference to the
> following points:
>
> 1) SMS.
> 2) gtalk and other social media.
> 3) GUI.
> 4) Any main difference?
>
> Regards
> Bilal
>
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> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] Sip and the media path

[David Wessell]

We're running asterisk 1.8 in the DC on a public IP address.

Connecting to it are about 200 phones behind a LAN in a remote location.

Is there a way to reliably keep asterisk out of the media stream on internal 
calls inside that LAN? All phones are Polycom Soundpoint phones.

Asterisk would say in the media stream for any calls that traverse from LAN to 
WAN. However it would step out for LAN to LAN calls.

Thanks
David

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[asterisk-users] Asterisk 1.8 and 11

[bilal ghayyad]

Hello;

How I can compare between Asterisk 1.8 and 11 with reference to the following 
points:

1) SMS.
2) gtalk and other social media.
3) GUI.
4) Any main difference?

Regards
Bilal

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