MYSQL INSERT QUESTION IN DIALPLAN

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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…

Asterisk Users 3.4 years ago 5 Answers

syntax error from digium fax manual ??

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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)})

Asterisk Users 3.4 years ago 1 Answer

Combining multiple SIP providers

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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…

Asterisk Users 3.4 years ago 1 Answer

ChannelRedirect with callee channel

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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…

Asterisk Users 3.4 years ago 0 Answers

Call Deflection with DAHDISendCallreroutingFacility

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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…

Asterisk Users 3.4 years ago 1 Answer

Monitoring voice-quality with sip/rtp/rtcp

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After some customer complaints I find myself tcpdumping, gzipping and
transferring large packagedumps over the network to be analyzed. While this manual process isn't a long-term solution, I'm evaluating
different options. Aside from the manual thing I could see two variants:
- Dump the traffic (on the server or another via switch port
mirroring/monitoring) and analyze it with tshark
- Analyze the traffic in asterisk How do you monitor call quality for you services? (Right now I use
asterisk 1.4, but are in the process to migrate to 1.8 or 10.) I looking

Asterisk Users 3.4 years ago 4 Answers