Re: [asterisk-users] New to Asterisks, Couple of Questions

2014-09-05 Thread Shishir Pokharel
Start from
http://www.voip-info.org/
or
Asterisk : The Future of Telephony Book


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Horace Miles
Sent: Friday, September 05, 2014 12:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] New to Asterisks, Couple of Questions

Hello everyone, my name is Miles, I am fairly new to asterisk.  I have recently 
begun to learn asterisk and I have a couple of questions.


1.After installing asterisk using the following instructions;

a.  sudo mkdir /usr/src/asterisk && cd /usr/src/asterisk

b.  sudo wget 
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-11.2.1.tar.gz

c.  sudo tar -xvzf asterisk-11.2.1.tar.gz

d.  cd ./asterisk-11.2.1

e.  sudo make clean

f.  sudo ./configure

g.  sudo make

h.  sudo make install

i.  sudo make samples

j.  sudo make config

k.  sudo service asterisk start
The astDB did not automatically initialize.  i.e. performing the following 
commands

Asterisk -rx "database show" produced no output.

IS THERE A WAY TO INITIALIZE THIS DATABASE  WITHOUT HAVING TO INSERT A FAMILY 
AND VALUE?


2.Does any one know of a simple open source switchboard script  that 
places the caller into a que and gives CALLERID, TIME ON HOLD, and a way to 
either forward the call or answer it?  Something I use to modify and learn how 
it all works more or less?
Thanks ahead of time.

Miles

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Re: [asterisk-users] unidata incom ICW-1000G - On asterisk

2014-09-05 Thread George Joseph
On Fri, Sep 5, 2014 at 7:26 AM, Bryant Zimmerman  wrote:

> I am trying to use an ICW-1000G wireless handset connected to an asterisk
> server remotely
>
> The user is working from an offsite location and it appears that the
> device is not sending out keep-alives or stun.
> The manufacture is not being of assistance at all.  I am wondering if
> anyone has worked with these units or has any ideas of what I could do to
> make them work.
>
> Anyone have a better alternative. The units must support wifi, be able to
> clip on belt and support an external headset (wired is fine) Battery life
> should support all day use in a standard biz env. Work thru a firewall from
> offsite.
>
> The ICW-1000G supports all of these requirements except the Work thru a
> firewall from off-site.
>
> If the MFG can't get a fix or someone does not have any ideas I would
> recommend people stay away from the ICW-1000G if you have to use them
> off-site or for hosted connections.
>
> Thanks
> Bryant
>
>

I use the older Unidata WPU-7800s remotely with both Asterisk and the phone
behind firewalls with no problems.   What are your symptoms and Asterisk
settings for the peer?
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Re: [asterisk-users] unidata incom ICW-1000G - On asterisk

2014-09-05 Thread Bryant Zimmerman
I am trying to use an ICW-1000G wireless handset connected to an asterisk 
server remotely
  
 The user is working from an offsite location and it appears that the 
device is not sending out keep-alives or stun.
 The manufacture is not being of assistance at all.  I am wondering if 
anyone has worked with these units or has any ideas of what I could do to 
make them work.
  
 Anyone have a better alternative. The units must support wifi, be able to 
clip on belt and support an external headset (wired is fine) Battery life 
should support all day use in a standard biz env. Work thru a firewall from 
offsite.
  
 The ICW-1000G supports all of these requirements except the Work thru a 
firewall from off-site.
  
 If the MFG can't get a fix or someone does not have any ideas I would 
recommend people stay away from the ICW-1000G if you have to use them 
off-site or for hosted connections.
  
 Thanks
 Bryant

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Re: [asterisk-users] New to Asterisks, Couple of Questions

2014-09-05 Thread Joshua Colp

Horace Miles wrote:





Asterisk –rx “database show” produced no output.

IS THERE A WAY TO INITIALIZE THIS DATABASE WITHOUT HAVING TO INSERT A
FAMILY AND VALUE?


Initialize it with what? If Asterisk has started and no previous 
database existed it has created a new one with the schema embedded in 
the code. Since usage of the astdb is up to everything else there is 
nothing stored in it until other stuff puts something there.


Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk with PJSIP

2014-09-05 Thread Joshua Colp

エムディーシー太郎 wrote:

Hi All,

I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code
on CentOS7.
--https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject






--
2. dial from 9001 to 9002

*CLI> -- Executing [9002@internal:1] Dial("PJSIP/9001-",
"PJSIP/9002,20") in new stack
 -- Called PJSIP/9002
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Auto fallthrough, channel 'PJSIP/9001-' status is
'CHANUNAVAIL'


What is shown if you do "pjsip set logger on" and then try to place the 
call?


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asterisk with PJSIP

2014-09-05 Thread Rainer Piper

Hi,

can you check the Linphone Extension 9002!!

The port is missing!
Contact:  9002/sip:9002@192.168.177.189 
:
Avail  24.210


Regards
Rainer

Am 05.09.2014 um 11:55 schrieb エムディーシー太郎:

Hi All,

I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code 
on CentOS7.

--https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject

The installation is OK.
But the connected SIP cilents (both Linphone on Windows7) cannot 
communicate.


I hope your comment such as the testing for resolving the problem.

My status is the following(1 and 2).
Why 'Everyone is busy/congested at this time (1:0/0/1)'?
(1:0/0/1<---num.nochan is 1.)

--
1. endpoint
*CLI> pjsip show endpoints
 Endpoint:  
  
I/OAuth: 

Aor:  

  Contact:  
  
  Transport:

   Identify: 

Channel:  
  

Exten:   CLCID: 
 
=
 Endpoint: 9001 Not in 
use0 of inf

 InAuth:  auth9001/9001
Aor: 9001  10
  Contact:  9001/sip:9001@192.168.177.180:16060 
 Avail  25.048
  Transport:  transport-udp udp  0  0 0.0.0.0:5060 

 Endpoint: 9002 Not in 
use0 of inf

 InAuth:  auth9002/9002
Aor: 9002  10
*  Contact:  9002/**sip:9002@192.168.177.189 
**Avail  24.210*
  Transport:  transport-udp udp  0  0 0.0.0.0:5060 



--
2. dial from 9001 to 9002

*CLI> -- Executing [9002@internal:1] Dial("PJSIP/9001-", 
"PJSIP/9002,20") in new stack

-- Called PJSIP/9002
  == Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/9001-' status is 
'CHANUNAVAIL'

--

Thanks,
MMEEGGAA






--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: +49 228 97167161 
P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test)
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[asterisk-users] Asterisk with PJSIP

2014-09-05 Thread エムディーシー太郎
Hi All,

I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on
CentOS7.
--
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject

The installation is OK.
But the connected SIP cilents (both Linphone on Windows7) cannot
communicate.

I hope your comment such as the testing for resolving the problem.

My status is the following(1 and 2).
Why 'Everyone is busy/congested at this time (1:0/0/1)'?
(1:0/0/1<---num.nochan is 1.)

--
1. endpoint
*CLI> pjsip show endpoints
 Endpoint:  
  
I/OAuth:

Aor:  

  Contact:  
  
  Transport:

   Identify:

Channel:  
  
Exten:   CLCID: 
 
=
 Endpoint:  9001 Not in
use0 of inf
 InAuth:  auth9001/9001
Aor:  9001  10
  Contact:  9001/sip:9001@192.168.177.180:16060
Avail  25.048
  Transport:  transport-udp udp  0  0  0.0.0.0:5060
 Endpoint:  9002 Not in
use0 of inf
 InAuth:  auth9002/9002
Aor:  9002  10
  Contact:  9002/sip:9002@192.168.177.189
Avail  24.210
  Transport:  transport-udp udp  0  0  0.0.0.0:5060

--
2. dial from 9001 to 9002

*CLI> -- Executing [9002@internal:1] Dial("PJSIP/9001-",
"PJSIP/9002,20") in new stack
-- Called PJSIP/9002
  == Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/9001-' status is
'CHANUNAVAIL'
--

Thanks,
MMEEGGAA
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[asterisk-users] New to Asterisks, Couple of Questions

2014-09-05 Thread Horace Miles
Hello everyone, my name is Miles, I am fairly new to asterisk.  I have
recently begun to learn asterisk and I have a couple of questions.

 

1.After installing asterisk using the following instructions;

a.  sudo mkdir /usr/src/asterisk && cd /usr/src/asterisk

b.  sudo wget
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1
1.2.1.tar.gz

c.  sudo tar -xvzf asterisk-11.2.1.tar.gz

d.  cd ./asterisk-11.2.1

e.  sudo make clean

f.  sudo ./configure

g.  sudo make

h.  sudo make install

i.  sudo make samples

j.  sudo make config

k.  sudo service asterisk start

The astDB did not automatically initialize.  i.e. performing the
following commands 



Asterisk -rx "database show" produced no output.

 

IS THERE A WAY TO INITIALIZE THIS DATABASE  WITHOUT HAVING TO INSERT A
FAMILY AND VALUE?

 

2.Does any one know of a simple open source switchboard script
that places the caller into a que and gives CALLERID, TIME ON HOLD, and
a way to either forward the call or answer it?  Something I use to
modify and learn how it all works more or less?

Thanks ahead of time.

 

Miles

 

-- 
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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   http://www.asterisk.org/hello

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