* You are viewing the archive for April 9th, 2012

MYSQL INSERT QUESTION IN DIALPLAN

I am not a programmer and I have learned so much from examples and the list.
Perhaps someone could tell me what I am doing wrong in my example below:

I am getting the caller ID and caller name from my local POTS line and I
want to add it into a sql table. I am trying with the following code but
the data never gets put into the table.

Can anyone correct my syntax and tell me what I am doing wrong?

[callerinfo]
exten => s,1,MYSQL(Connect connid localhost myuser mypassword cnam)
exten => s,n,MYSQL(Query resultid ${connid} INSERT INTO `calleridcapture`
(`number`,`name`) VALUES (${CALLERID(num)},${CALLERID(name)})
exten => s,n,MYSQL(Clear ${resultid})
exten => s,n,MYSQL(Disconnect ${connid})
exten => s,n,NoOp(Callerid Name ${CALLERID(name)})
exten => s,n,NoOp(Callerid Number ${CALLERID(num)})

The NoOP does show the correct CALLERID name & number when I test it. The
information just doesn’t go into my calleridcapture table in the cnam
database.

Thanks very much for your help
Again I am not a programmer and I am sure my syntax is wrong.

This is Asterisk 1.8.10.0

syntax error from digium fax manual ??

I’ve cut and pasted from the digium fax admin manual:

exten => send,1,NoOp(**** SENDING FAX ****)
exten => send,n,Wait(6)
exten => send,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten => send,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})

Combining multiple SIP providers

Simply modify your dialplan to do this:

Exten => s,1,dial(SIP/${ARG1}@SIP1&SIP/${ARG1}@SIP2..)

If you are using a “cookbook” dialplan, it’s probably a 4 or 5 line change.

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Anita Hall
Sent: Monday, April 09, 2012 9:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Combining multiple SIP providers

Hi

What is the best way to combine multiple SIP providers to achieve

1) Higher concurrency (for eg. 2 providers with 50 concurrent calling limits
could be combined to give a limit of 100)
2) Redundancy (use another if one is down)

I have a feeling that this will need some SIP Proxy like OpenSIPS but what
could be the architecture ?

Much thanks!

VMWI DAHDI

ChannelRedirect with callee channel

Hello List,

I tried 3way conferencing:
http://lists.digium.com/pipermail/asterisk-users/2011-June/263345.html
Asterisk Version 1.8.11.
A calls B and A press **0 to invoke ‘3way-start’ feature. B is
redirected correctly by ChannelRedirect. B talks to C and starts 3-way
conference. All is working.
But: A calls B and now B press **0. Dialplan calls ChannelRedirect and
tries to redirect A into conference room, but it is not working. Seems
the command is waiting for something. B talks to C and press **1 to
start the conference. B and C are in conference room and can talk.
Channel of A hang in the meantime. No music on hold, nothing. After B
and C hangs up the ChannelRedirect starts working and A gets into the
conference room. That’s useless…
Someone with any hints? For which reason does the Channelredirect for
channel A wait for B being hung up?

Call Deflection with DAHDISendCallreroutingFacility

Hi
I want to use Call Deflection with DAHDISendCallreroutingFacility Application.
I use Asterisk:1.8.11 libpri:1.4.12 facilityenable=yes transfer=yes
 my dialplan is like this:

[Call-Deflection]
exten => 66,n,Proceeding()
exten => 66,1,wait(5)
exten => 66,n,DAHDISendCallreroutingFacility(88050048,8262000,cfb)
exten => 66,n,wait(5)
exten => 66,n,Hangup()

after Executing DAHDISendCallreroutingFacility(“DAHDI/i1/2188602827-3″, “88050048,8262000,cfb”)
in new stack  == Spawn extension (Call-Deflection, 66, 3) exited non-zero on ‘DAHDI/i1/2188602827-3′

Asterisk exit immediately and last wait(5) won’t Excute.

I used another PRI Analyzer and this is message sequence:

Asterisk    < --setup--    Local exchange
Asterisk  –proceeding–> 
Local exchange
Asterisk   –facility–>  Local exchange
Asterisk   –Disconnect(Subscriber Absent)–> Local exchange
Asterisk    < --Release-- Local exchange
Asterisk    –Release complete–> Local exchange

from the Analyzer report Asterisk send Disconnect immediately after Facility message
(don’t wait for response from Local exchange).
please help me solve this problem

Regards
M.Shirazi