[asterisk-users] Question about SIP warning

[CDR]

I get tons of these messages
chan_sip.c:10088 process_sdp: Declining non-primary audio stream:
audio 30660 RTP/AVP 4 101 13
What does it mean and does it show a problem like one-way audio?
Thanks for your help.

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Re: [asterisk-users] Failover / modifying response time

[Johan Wilfer]

Den 2014-09-04 18:05, Stephen More skrev:

I was able to get a packet trace of this event

Time
312.353549 - INVITE to primary
313.222303 - INVITE to primary ( suspected resend of frame )
314.289215 - INVITE to backup
315.397120 - INVITE to backup ( suspected resend of frame )

So is primary just too slow to answer ? I am not seeing anything in the
logs on primary.



You can try with Wait(2) to wait two seconds before you do Answer(). 
This will delay the response to the INVITE with two seconds.


You don't want to wait to long thought, so maybe you can test your way 
to something that works for you.


Good luck!


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Johan Wilfer

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

[Carlos Chavez]

On 9/5/2014 2:18 AM, Horace Miles wrote:


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 amp;amp; 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.




Actually, if you are looking for a prebuilt ready to use Asterisk 
your best bet is to download a distribution like Elastix. If you want to 
start from scratch without any experience you will be hitting to many walls.
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