Re: [asterisk-users] Asterisk 11 & GoogleVoice/Motif

[Chris Gentle]

Awesome, thanks.  I'll give it a try.
On Mar 11, 2013 4:56 PM, "Joshua Colp"  wrote:

> Chris Gentle wrote:
>
>> I'm currently running Asterisk 11.2.1 and I've noticed that when
>> asterisk has been up for a while (usually about a day), outgoing calls
>> through GoogleVoice fail to complete.  I hear it ringing on my end but
>> the caller never hears the phone ring.  A simple restart of Asterisk
>> seems to clear it up for another day or so.  Has anyone else noticed this?
>>
>
> This is fixed in Asterisk 11.3.0-rc1. The Google XMPP server has become
> prone to disconnecting as of late, which triggers a bug in older versions
> where chan_motif ignores XMPP traffic.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
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Re: [asterisk-users] Serviced Office operator panel

[Andrew Yager]

Hi,

It's a great console, should have included it in my list. 

Sadly doesn't meet SO requirements. :(

Andrew

On 12/03/2013, at 11:00 AM, Patrick Lists  
wrote:

> On 03/12/2013 12:07 AM, Andrew Yager wrote:
>> Hi,
>> 
>> I'm trying to find (with some desperation now) a decent web based or
>> application based UI that integrates with an Asterisk based PBX and is
>> designed for a Serviced Office environment.
>> 
>> Key features we're looking for:
> 
> Don't know if it covers your requirements but here's another commercial 
> solution: http://www.getisymphony.com/
> 
> Regards,
> Patrick
> 
> 
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Re: [asterisk-users] Serviced Office operator panel

[Patrick Lists]

On 03/12/2013 12:07 AM, Andrew Yager wrote:

Hi,

I'm trying to find (with some desperation now) a decent web based or
application based UI that integrates with an Asterisk based PBX and is
designed for a Serviced Office environment.

Key features we're looking for:


Don't know if it covers your requirements but here's another commercial 
solution: http://www.getisymphony.com/


Regards,
Patrick


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Re: [asterisk-users] Serviced Office operator panel

[Andrew Yager]

On 12/03/2013, at 10:15 AM, Chris Bagnall  wrote:

> On 11/3/13 11:07 pm, Andrew Yager wrote:
>> Basically, if you know of a product, open or closed source, and would like 
>> to sell it to me and you think it does the job, or you've seen something 
>> that works, contact me off list ASAP!
> 
> Actually, please post *on* the list if you know or have used something that 
> meets the above. I suspect many of us would find such a product or 
> application useful from time to time.

Certainly planning to post back; just imagined that I would elicit a series of 
"hey - I can sell you this" emails. I'm certainly happy for discussion of Open 
Source integrations/options on-list!

Andrew
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Re: [asterisk-users] Serviced Office operator panel

[Chris Bagnall]

On 11/3/13 11:07 pm, Andrew Yager wrote:

Basically, if you know of a product, open or closed source, and would like to 
sell it to me and you think it does the job, or you've seen something that 
works, contact me off list ASAP!


Actually, please post *on* the list if you know or have used something 
that meets the above. I suspect many of us would find such a product or 
application useful from time to time.


Kind regards,

Chris
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[asterisk-users] Serviced Office operator panel

[Andrew Yager]

Hi,

I'm trying to find (with some desperation now) a decent web based or 
application based UI that integrates with an Asterisk based PBX and is designed 
for a Serviced Office environment.

Key features we're looking for:

Concept of a client with multiple staff/associates
Recognition of inbound DIDs with display of company details and greeting
Support for multiple receptionists
Some concept or mechanism for charging
Visibility of extension status (i.e. is someone on the phone)
API, open code, or something that allows us to have users set presence 
information from the handset
Some sort of Web UI or similar, or API to allow users to update their 
information

I know it's a big ask, but I'm really hoping someone else has tried to solve 
this problem before now. We've currently played with FOP, Noojee Reception 
Console, Hosted Suite… all of which are good for what they do, but here is my 
take:

FOP2
Good for single office
Clear visibility, but becomes unwieldy with a large number of extensions
No way to group extensions into clients, no way to display additional member 
information

Hosted Suite
Out of the box ticks all the requirements but:
Does not support BLF display
Impossible to drag billing information out
Commercial Support Team (given it's a closed source product) seem to not know 
the product very well
Sales staff are really good at copying and pasting from the wiki, but not good 
at responding to feature or development requests

Noojee Reception Console
Old Java based app
Difficult to get running
Possible the best "compromise solution"
Unsupported (really) with no development activity
Doesn't really have the right concept of a serviced office.

Basically, if you know of a product, open or closed source, and would like to 
sell it to me and you think it does the job, or you've seen something that 
works, contact me off list ASAP!

Thanks,
Andrew

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Re: [asterisk-users] Asterisk 11 & GoogleVoice/Motif

[Joshua Colp]

Chris Gentle wrote:

I'm currently running Asterisk 11.2.1 and I've noticed that when
asterisk has been up for a while (usually about a day), outgoing calls
through GoogleVoice fail to complete.  I hear it ringing on my end but
the caller never hears the phone ring.  A simple restart of Asterisk
seems to clear it up for another day or so.  Has anyone else noticed this?


This is fixed in Asterisk 11.3.0-rc1. The Google XMPP server has become 
prone to disconnecting as of late, which triggers a bug in older 
versions where chan_motif ignores XMPP traffic.


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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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[asterisk-users] Asterisk 11 & GoogleVoice/Motif

[Chris Gentle]

I'm currently running Asterisk 11.2.1 and I've noticed that when asterisk
has been up for a while (usually about a day), outgoing calls through
GoogleVoice fail to complete.  I hear it ringing on my end but the caller
never hears the phone ring.  A simple restart of Asterisk seems to clear it
up for another day or so.  Has anyone else noticed this?

-- 
Chris
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Re: [asterisk-users] digium card and virualbox

[Kevin Larsen]

From:   Hans Witvliet 
To: asterisk-users@lists.digium.com, 
Date:   03/11/2013 03:00 PM
Subject:Re: [asterisk-users] digium card and virualbox
Sent by:asterisk-users-boun...@lists.digium.com


I am not mixing. I need this for LAB testing. 
How? This PCI passthrough, how to enable it on virualbox?
---

http://www.virtualbox.org/manual/ch09.html

First link after googling "virtualbox pci passthrough"
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Re: [asterisk-users] digium card and virualbox

[Hans Witvliet]

-Original Message-
From: bilal ghayyad 
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion

To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] digium card and virualbox
Date: Sun, 10 Mar 2013 20:18:52 -0700 (PDT)

I am not mixing. I need this for LAB testing. 
How? This PCI passthrough, how to enable it on virualbox?
---
> > Hi All;
> >
> > How to let the virualbox (ubuntu OS) to be able to see
> the digium card? Because when I install elastix or asterisk
> with dahdi, it is not able to see the digium card if the
> installation though the virualbox .. What is the solution?
> The solution is to run Ubuntu and Asterisk on your hardware
> natively, 
> not through VirtualBox.
> 
> Virtualisation and high-performance hardware such as
> telephony cards  
> (it will be creating 8000 interrupts per phone line per
> second)  do not 
> mix, I am afraid.

What you might do, is  running an very elementary asterisk on the iron,
just acting as an PSTN-gateway. 
And run your experimental asterisk as a virtual client.

HW


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Re: [asterisk-users] Sending SMS from asterisk

[Asghar Mohammad]

hi,
i have not made proper patches but edit chan_mobile.c
here is detail:
tested with asterisk 1.6.0 and asterisk 1.6.2.

"mobile not connect with asterisk"

asterisk 1.6.0 line 275  is addr.rc_channel = (uint8_t) 1; change
to  addr.rc_channel = (uint8_t) 0;
asterisk 1.6.2.x line 1290  is addr.rc_channel = (uint8_t) 1; change
to  addr.rc_channel = (uint8_t) 0;
asterisk 1.6.2.x from line 2022

 if (buf[i] == ',') {
state++;
}
case 1: /* find the opening quote (") */
if (buf[i] == '"') {
state++;
}
case 2: /* mark the start of the number */

change to


 if (buf[i] == ',') {
state++;
}
break;
case 1: /* find the opening quote (") */
if (buf[i] == '"') {
state++;
}
break;
case 2: /* mark the start of the number */


note 2 break after every statement.

On Mon, Mar 11, 2013 at 7:40 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 03/11/2013 07:07 PM, Asghar Mohammad wrote:
>
>> HI Bilal,
>> i am using chan_mobile for call termination, you can use it but you need
>> to tweak chan_mobile.c it is broken from a long time.
>> let me know if you want give it a try.
>>
>
> If you could send the patches you made to chan_mobile to this mailing list
> then other Asterisk users can benefit from your work and use chan_mobile
> too.
>
> Regards,
> Patrick
>
>
>
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Re: [asterisk-users] Sending SMS from asterisk

[Patrick Lists]

On 03/11/2013 07:07 PM, Asghar Mohammad wrote:

HI Bilal,
i am using chan_mobile for call termination, you can use it but you need
to tweak chan_mobile.c it is broken from a long time.
let me know if you want give it a try.


If you could send the patches you made to chan_mobile to this mailing 
list then other Asterisk users can benefit from your work and use 
chan_mobile too.


Regards,
Patrick


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[asterisk-users] compiling imap

[Dagoberto Ramirez Gaxiola]

HI



I have a problem compiling asterisk with imap



When I enter this command



*sudo ./configure
--with-imap=/usr/src/asterisk-complete/thirdparty/imap-2007f/*

* *

at the end I allways receive this error



*asterisk the IMAP TK installation appears to be missing or broken*

* *

i have looked a lot in the internet but I still cannot find the answer

* *

[image: dagoberto_ramirez]

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this communication, or the use of its contents, is prohibited. If you have
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message de tous les systèmes de stockage de données. Je vous remercie.
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Re: [asterisk-users] Sending SMS from asterisk

[Asghar Mohammad]

HI Bilal,
i am using chan_mobile for call termination, you can use it but you need
to tweak chan_mobile.c it is broken from a long time.
let me know if you want give it a try.

On Mon, Mar 11, 2013 at 6:22 PM, bilal ghayyad  wrote:

> -
> > > What are the elements of this solution? Is it only: 3G
> > dongles and chan_dongle only? Or there are something else?
> >
> > Bash and perl programing, asterisk and chan_dongle.
> >
>
> * Bash and perl programing to do what? It is going to use AMI instead of
> sending the messages from the commands given in the extensions.conf?
>
> Why to use chan_dongle and not chan_mobile?
>
> Regards
> Bilal
>
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Re: [asterisk-users] Laptop error

[termo termosel]

Ahhh ok, thanks!
 > Date: Mon, 11 Mar 2013 14:34:32 +0100
> From: asterisk-l...@puzzled.xs4all.nl
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Laptop error
> 
> On 03/11/2013 12:53 PM, termo termosel wrote:
> > Hi,
> >
> > I have Ubuntu and Asterisk 11.2.1 in a boot USB. When I put it in
> > desktop computer, asterisk starts without problem but if I insert the
> > same USB in a laptop computer Asterisk doesn't start. Is it possible
> > because different microprocessors?
> 
> Yes. If you made the USB stick on a x86_64 (64 bit) computer and then 
> try it on a x86 (32 bit) laptop, it will not work.
> 
> Regards,
> Patrick
> 
> 
> 
> 
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Re: [asterisk-users] Sending SMS from asterisk

[bilal ghayyad]

-
> > What are the elements of this solution? Is it only: 3G
> dongles and chan_dongle only? Or there are something else?
> 
> Bash and perl programing, asterisk and chan_dongle.
> 

* Bash and perl programing to do what? It is going to use AMI instead of 
sending the messages from the commands given in the extensions.conf?

Why to use chan_dongle and not chan_mobile?

Regards
Bilal

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Re: [asterisk-users] Laptop error

[Patrick Lists]

On 03/11/2013 12:53 PM, termo termosel wrote:

Hi,

I have Ubuntu and Asterisk 11.2.1 in a boot USB. When I put it in
desktop computer, asterisk starts without problem but if I insert the
same USB in a laptop computer Asterisk doesn't start. Is it possible
because different microprocessors?


Yes. If you made the USB stick on a x86_64 (64 bit) computer and then 
try it on a x86 (32 bit) laptop, it will not work.


Regards,
Patrick




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Re: [asterisk-users] digium card and virualbox

[Patrick Lists]

On 03/11/2013 04:18 AM, bilal ghayyad wrote:

I am not mixing. I need this for LAB testing.
How? This PCI passthrough, how to enable it on virualbox?


It's in the VirtualBox manual.

Regards,
Patrick


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Re: [asterisk-users] Digium Wildcard TDM800P not working with DAHDI

[Gilberto Sanches]

Ow okayI'll use your instructions. Hopefully it works. Thanks by the
way :)


On Mon, Mar 11, 2013 at 6:17 AM, Doug Lytle  wrote:

> A J Stiles wrote:
>
>> Have you tried building LibPRI, DAHDI and Asterisk from Source Code
>>
>
> Just a note, newer versions of libpri now require dahdi to be compiled
> first.
>
> 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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[asterisk-users] Laptop error

[termo termosel]

Hi,

I have Ubuntu and Asterisk 11.2.1 in a boot USB. When I put it in desktop 
computer, asterisk starts without problem but if I insert the same USB in a 
laptop computer Asterisk doesn't start. Is it possible because different 
microprocessors?

The error is: instrucción ilegal ('core generado)

Thanks!
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Re: [asterisk-users] IPv6 and IPv4 binding address on a server with 2 network cards

[Miguel Baptista]

Hi Asghar,

Thanks for you reply. Which Asterisk version are you using?

I am using Asterisk 11.1.0
when I use the /bindaddr  /parameters with specific IP addresses,
Asterisk will listen only on the last entry.

For example, when I have
/bindaddr=ipv4A:port
//bindaddr=[ipv6A]:port /

it will listen only on the IPv6A address

and when I have the other way around:
/
bindaddr=[ipv6A]:port /
/bindaddr=ipv4A:port
//
/Asterisk will only listen on the IPv4A address.

The only way I found to force asterisk to listen on both IPv4A and IPv6
A was to use/bindaddr=[::] /but it makes asterisk to listen also on the
other IP addresses.

Maybe this is fix on a newer Asterisk version.

- Miguel Baptista

On 3/10/2013 8:04 PM, Asghar Mohammad wrote:
> hi,
> i am using similer setup just put / bindaddr=ipv4A:port
> and //bindaddr=[ipv6A]:port ans it should work./
>
> On Sun, Mar 10, 2013 at 3:04 PM, Miguel Baptista
> mailto:miguel.bapti...@uninett.no>> wrote:
>
> Hello,
>
> I am doing some tests with asterisk on a dual-stack environment. 
> I have some doubts regarding asterisk binding addresses on a
> server with 2 network cards.
>
> According to asterisk documentation:
>
> /; With the current situation, you can do one of four things:/
> /;  a) Listen on a specific IPv4 address.  Example:
> bindaddr=192.0.2.1/
> /;  b) Listen on a specific IPv6 address.  Example:
> bindaddr=2001:db8::1/
> /;  c) Listen on the IPv4 wildcard.Example:
> bindaddr=0.0.0.0/
> /;  d) Listen on the IPv4 and IPv6 wildcards.  Example:
> bindaddr=::/
> /; (You can choose independently for UDP, TCP, and TLS, by
> specifying different values for/
> /; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)/
> /; (Note that using bindaddr=:: will show only a single IPv6
> socket in netstat./
> /;  IPv4 is supported at the same time using IPv4-mapped IPv6
> addresses.)/
> /;/
> /; You may optionally add a port number. (The default is port
> 5060 for UDP and TCP, 5061/
> /; for TLS)./
> /;   IPv4 example: bindaddr=0.0.0.0:5062 /
> /;   IPv6 example: bindaddr=[::]:5062/
> /;/
> /; The address family of the bound UDP address is used to
> determine how Asterisk performs/
> /; DNS lookups. In cases a) and c) above, only A records are
> considered. In case b), only/
> /;  records are considered. In case d), both A and 
> records are considered. Note,/
> /; however, that Asterisk ignores all records except the first
> one. In case d), when both A/
> /; and  records are available, either an A or  record
> will be first, and which one/
> /; depends on the operating system. On systems using glibc,
>  records are given/
> /; priority./
>
>
> Lets say that I have two network cards: A and B.
> Both interfaces have IPv4 and IPv6 addresses: IPv4 A, IPv6 A, IPv4
> B and IPv6 B.
>
> How can I make asterisk to run only on B network addresses (IPv6
> and IPv4)? The /bindaddr=[::] /config parameter tells asterisk to
> run on all available addresses, including the addresses on the A
> network. But that's not exactly what I want.
>
> - Miguel Baptista
>
>
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>
>
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Re: [asterisk-users] digium card and virualbox

[Doug Lytle]

bilal ghayyad wrote:

I am not mixing. I need this for LAB testing.
How? This PCI passthrough, how to enable it on virualbox?




Google is your friend:

http://www.virtualbox.org/manual/ch09.html#pcipassthrough

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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Re: [asterisk-users] Digium Wildcard TDM800P not working with DAHDI

[Doug Lytle]

A J Stiles wrote:

Have you tried building LibPRI, DAHDI and Asterisk from Source Code


Just a note, newer versions of libpri now require dahdi to be compiled 
first.


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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Re: [asterisk-users] Digium Wildcard TDM800P not working with DAHDI

[A J Stiles]

On Sunday 10 March 2013, Gilberto Sanches wrote:
> I'm working for over 4 months now to fix this problem. The machine specs:
> 
> On a P4 2.8Ghz, 1.5 GB RAM, 40 Gb HDD I installed:
> 
> - I've tried AsteriskNOW 2.02
> - Installed CentOS 6.3 and then compiled Asterisk 11, Libpri 1.4 and DAHDI
> 2.6
> - Installed PBX in a Flash - Purple 20631
> After the installations I used these commands:
> 
> - dahdi_genconf
> - dahdi_cfg -v
> - /etc/init.d/dahdi restart
> - /etc/init.d/asterisk restart
> - shutdown -r now
> 
> Created and configured: Extensions, Trunk, inbound route, outbound route.
> rebooted the system and then went into Asterisk console and check if
> channels are configured with these commands:
> 
> 
> - dahdi show status
> 
> - dahdi show channels

Have you tried building LibPRI, DAHDI and Asterisk from Source Code  (in that 
order)?  Maybe you have something missing that the package maintainer forgot 
to mention as a dependency  (it does happen sometimes).  A build from Source 
Code will prove beyond all reasonable doubt that you have nothing essential 
missing.

Also, running dahdi_genconf on its own isn't enough; it generates a 
dahdi_channels.conf for you, but you still have to paste this into your 
chan_dahdi.conf  (or add an include line).


-- 
AJS

Answers come *after* questions.

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