* You are viewing Posts Tagged ‘connection’

H323-sip: One Way Connection

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]
host2.168.0.146
type=friend context=from-trunk


[to-146]
type=peer host2.168.0.146
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

Round-robin In Asterisk 1.4

I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210
with 2 port E1.

now i bought another card Diguim TE410 and I want to add it

the current configuration : connection (WIMAX) from the first ISP and connection (fiber optic) from the secend ISP.

the desired configuration : connection (WIMAX) and connection (radio beam)
from the first ISP.from the second ISP no change (still have the fibre optic)

my question how to active the round-robin in asterisk 1.4 in order to active the 3 technology (WIMAX-radio beam and fibre optic)
any help please

Remote SIP Extension Best Practices

What are best practices for allowing connection by remote SIP
extensions over the internet? I’m thinking of putting the SIP inside a VPN connection.

Kind Regards, Chris

Enable CDR Logging?

Hello,

I am currently trying to set up CDR logging. I got all the ODBC stuff for my mysql server set up, also checked there’s a connection using “odbc show”
in the Asterisk console:

Name: asterisk


I also created a table for the CDR logs:


This table is also referenced in /etc/asterisk/cdr_adaptive_odbc.conf.

Now, what is the next step?
1) Do I have to call any CDR function in my dialplan for the CDR logging to become really active?
2) Do I have to create more fields in my asterisk_cdr table manually? Guess it should all be done automatically? Would I have needed the ID field at all?

Thanks for any hint :-)


Best regards Stefan

sip show peers

I have a process that runs on a server and does a simple ‘asterisk -rx
“sup show peers’ > /tmp/peers”
and then looks for any “(Unspecified)” items and reports them as having
lost connection.
My server is running 1.4.43 and the two boxes I am monitoring are also
running 1.4.43.
Once in a great while 1 of my boxes reports “(Unspecified)”. I am trying
to find out why.

How can I make the remote boxes have a shorter heart beat to checking
more frequently
with the server so as not to go “(Unspecified)”. By the time I log in
and check its already
back connected again.

Any other thoughts?

Thanks,

Jerry

Asterisk and the media path

I am attempting to get an asterisk server to step out of the media
path, but am running into a brick wall. Can someone assist? Here’s my
setup..

Ultimate SIP Provider —> LCR Trunk (Asterisk 1.6) —-> PBX (Asterisk 1.8).

I am attempting to get the trunk to step out of the media stream.
There is no NAT involved, all machines have a public IP.

In the trunk’s sip.conf I have:

directmedia=yes
directrtpsetup=yes

And on the connection to the pbx I have canreinvite=yes

On the pbx I have the trunk connection set to canreinvite=yes.

In the CLI on the LCR trunk I see: