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Asterisk on Debian - Help with initial setup

Posted on 2014-04-02
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Last Modified: 2014-04-03
Hi Experts,

I'm currently following Asterisk The Definitive Guide for my setup but can't seem to get a test call to *43.

This is what I have done so far:

1. On a Debian wheezy machine I entered:
    apt-get install asterisk

2. Moved sip.conf and extensions.conf to backup/

3. Edit sip.conf as follows:

[general]
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=no

[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=force_rport,comedia
dtmfmode=auto
disallow=all
allow=g722
allow=ulaw
allow=alaw


[1001](office-phone)
secret=1234


[1002](office-phone)
secret=1234

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4. Skip the bit about iax.conf

5. Edit entensions.conf as follows:

[LocalSets]
exten => 1001,1,Dial(SIP/1001)
exten => 1002,1,Dial(SIP/1002)

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6. Start asterisk with:

asterisk -c

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7. Setup Blink softphone with the following:

Display name: 1001
Sip address: 1001@192.168.0.222 (have setup static ip for this asterisk box)
Password: 1234

8. I am getting the following messages from CLI:

NOTICE[2930]: chan_sip:24850 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1001

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9 After performing a fail test call to *43 I get the following from CLI:
WARNING[2930]: chan_sip.c:9187 process_sdp: We are requesting SRTP, but they responded without it!

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Please help point me in the right direction.

Many Thanks
Ricky
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LVL 20

Accepted Solution

by:
José Méndez earned 2000 total points
ID: 39972746
Yo yo yo!!! My man Ricky!! Long time no see..... xD

Here my friend:

This line is not important for now, I will gladly let you know how to fix it when you get to the voicemail part, it is just Blink asking Asterisk if this account has pending voice messages (how cool huh!!??)

NOTICE[2930]: chan_sip:24850 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1001

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Now, you are saying that from Blink phones you are dialing *43? If so, I don't think that is going to work based on the dialplan chunk shown here, the only 2 extensions listed are 1001 and 1002 within the context LocalSets. If you wish to ring extension 1002 from 1001, try this:

exten => *43,1,Dial(SIP/1002)

Reload your dialplan (dialplan reload from CLI) and try again.
0
 

Author Comment

by:Ronniel Allan Castanito
ID: 39973742
Hi willlywilburwonka!! yeah long time no see... Haha..

The test dialing to *43 is the echo test I was trying to make similar to the what I did in FreePBX.

 I thought *43 was an echo test universal to Asterisk or have I mistaken and *43 is only specific to FreePBX?

When I dial 1002 to test a call from 1001, the system doesn't ring but cuts off right away.

I have also tried changing the dialplan to *43 as suggested but with the same result.
0
 
LVL 20

Expert Comment

by:José Méndez
ID: 39973838
Ricky, lets see some debugs:

at the cli type:

core set debug 9...
core set verbose 9...

then call from 1002 from 1001 and copy the output. Would it be possible to attach the extensions.conf and sip.conf?
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Author Comment

by:Ronniel Allan Castanito
ID: 39973862
I get the same warning message when trying to dial 1002 from 1001:

WARNING[2930]: chan_sip.c:9187 process_sdp: We are requesting SRTP, but they responded without it!

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I've since changed my sip.conf and extensions.conf similar to this website:

http://draalin.com/basic-asterisk-configuration-in-ubuntu/

Should I change it back?
extensions.conf.jpg
sip.conf.jpg
blink.jpg
0
 

Author Comment

by:Ronniel Allan Castanito
ID: 39973878
Hmm.. I think I know what's wrong. When I performed the initial Asterisk install using the apt-get install method, I thought version 11 would be installed as per http://voipfreak.net/how-to-install-asterisk-on-debian/ comments on the bottom.

It turns out when typing "core show version" in CLI, I am told it's actually version 1.8.13.1.

I might have to go and do a fresh install from source.
0
 

Author Closing Comment

by:Ronniel Allan Castanito
ID: 39973994
Thanks willlywilburwonka for your input. I indeed installed the wrong version. Once I had correct version installed, test dial to 1002 from 1001 works!!
0
 
LVL 20

Expert Comment

by:José Méndez
ID: 39976041
Glad to hear that!

What was the new install method? Compilation?
0
 
LVL 20

Expert Comment

by:José Méndez
ID: 39976045
Oh, and what version of Asterisk did you install this time?
0
 

Author Comment

by:Ronniel Allan Castanito
ID: 39976853
Yeah thanks. :) I have version 11.8 installed now but I think the ATDG book has version 11.3 installed.

I just followed the book and downloaded the program from source using wget. The only problem I encountered was that Debian Wheezy didn't have a C++ compiler pre-installed. So I installed g++ package via aptitude before running the ./configure command and everything went smoothly from then on.

One question I did have regarding the ATDG book was that it added asteriskpbx user to the wheels group in sudo like so on page 44.

wheel:x:10:asteriskpbx

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Why would you do that? Wouldn't you create a more sensible group, say phonebox, and add the asteriskpbx user to there instead? Or am I missing the point and by adding to the wheel group it would increase security because that means it would be harder to guess the group that the asteriskpbx user is in?
0
 
LVL 20

Expert Comment

by:José Méndez
ID: 39977083
Those steps are meant for the asterisk process to be run as a non-root user. So they change the ownership of the /etc/asterisk folder among other steps to let that user open all necessary files and modules.

My asterisk box at home is also Debian based. Well, its a LinuxMint box to be honest, running as a server =S , so its Ubuntu really
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