H323-sip: One Way Connection

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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…

Asterisk Users 2.3 years ago 9 Answers

Round-robin In Asterisk 1.4

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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

Asterisk Users 2.5 years ago 2 Answers

Enable CDR Logging?

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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…

Asterisk Users 3 years ago 3 Answers

sip show peers

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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…

Asterisk Users 3.2 years ago 0 Answers

Asterisk and the media path

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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…

Asterisk Users 3.2 years ago 8 Answers

Question about asterisk to Cisco

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If I have a connection from Asterisk to Cisco Call Manager using SIP
can I send a text message using "SendText" from asterisk across the SIP
trunk to CCM and it convert that to text message to the skinny cisco phones? Does that work? Thanks, Jerry

Asterisk Users 3.4 years ago 1 Answer

AMI Originate double call

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Hi, probably is a problem already solved, but I cannot find a solution
anywhere.
so:
I tried to connect to Asterisk AMI using php and telnet, but the problem
is there anyway. 1. I call just 18 and a playback start.
2. then open a telnet connection and authenticate
3. originate one new call
4. two calls are originated ??? you could see it in the 4th and 5th line of asterisk cli:

Asterisk Users 3.4 years ago 0 Answers

ODBC connection does not reconnect after network interruption

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I've got an Asterisk 10.1.2 server using res_odbc to make a connection to a MSSQL server on a different machine for a timeclock extension we have running. The connection works just fine until Asterisk's network path to the MSSQL server gets disrupted (for whatever reason), and then all future attempts to contact the MSSQL server cause the timeclock call to hang, until Asterisk is completely restarted. Trying to module unload/load res_odbc.so crashes Asterisk. I've been using the following config, and tried with preconnect both on and off: [timeclock]
enabled => yes
dsn => timeclock
username => notauser

Asterisk Users 3.4 years ago 2 Answers

sip proxy

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hi all
how can i put a sip proxy; how it differs from asterisk pbx; currently
doing some test on asterisk; thru DSL connection the callers can hear only
one way;asterisk pbx is behind NAT; i am in search of a proper VOIP network
;appreciate some clues in this line
thanks -kbh

Asterisk Users 3.5 years ago 2 Answers