Asterisk unixODBC configuration files for MySQL and MariaDB

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1.0 Asterisk + unixODBC

Having almost all of our Asterisk configuration based on our preferred Database Management System is one of the greatest advantages that we have at the moment to deploy a VoIP solution. Now, having the possibility of building and integrated and unified communication solution in a non-intrusive way for client's company, while at the same time assuring scalability and flexibility, that's a mayor thing. That's precisely what we have at the moment of using Asterisk+unixODBC. unixODBC "allows the user or the system administrator to easily configure an application to use any ODBC compliant data source. This is…

Initial Configuration of Asterisk 4.2 years ago 0 Answers

Asterisk 1.6.1 Realtime SIP Users

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Hello,
I just implement the SIP Peers with MySQL. In the structure mySQL missing the following fields: nat = yes
notransfer = yes
dtmfmode = rfc2833
call-limit = 2
canreinvite = no
subscribecontext = blf subscribecontext (BLF) and call-limit (Protection) are very important ...
Can you help me? Best,
Mickael

Asterisk Users 4.2 years ago 2 Answers

cisco sip

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Hi
I have a working asterisk 1.8 system with 10 7961 phones. Everything works perfect except for the line ringing. I have 4 lines configured on each phone (IE 7002, 7022, 7042, and 7062). When I dial 7002 it rings line 1 like it should. But when I dial any of the other numbers it still rings only line 1 and not the actual line that it is assigned to. I checked the cli and it is dialing the correct number (7022, 7042 or 7062), but the phone still rings on 7002. Below is my .cnf.xml file for one…

Asterisk Users 4.2 years ago 0 Answers

not setting the fax header

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Hello, after I solved my problem with the fax processing after receiving,
I got another problem while sending a fax: the header is not set properly. I use a PHP_Script to upload a PDF file and to generate a call file. A bash script is looking for existent call files in the web directory
and moves
them the asterisk's outgoing directory. Ok, my call file looks like this: ======
$cf_commands = "Channel: local/$ext@commonn"
."MaxRetries: 5n"
."RetryTime: 60n"
."WaitTime: 60n"
."Context: faxn"
."Extension: 100n"
."Set: FAXFILE=$file_dstn"
."Set: FAXOPT(localstationid)=$_POST[mynumber]n"
."Set:…

Asterisk Users 4.2 years ago 0 Answers

asterisk recording problem

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Hi, I'm trying to record the incoming calls with Asterisk 1.6.2.15 and Dahdi
2.4.0. Without the line Record, everything works fine. If I add the line
Record, I can hear only a beep sound, the caller and the callee cannot
hear each other and there is no recording file. The extension in the extensions.conf:
SZERVERSZOBA=DAHDI/19,,rtT
exten => 57,1,Record(${TIMESTAMP}${CALLERID(num)}-${EXTEN}.wav)
exten => 57,n,Dial(${SZERVERSZOBA})
exten => 57,n,Playback(vm-nobodyavail)
exten => 57,n,Hangup() In the log:

Asterisk Users 4.2 years ago 1 Answer

Problem callerid ignored by using callfiles

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Hi all, i upgraded from 1.4.42 to 1.8.4.4. I generate a callfile with the option: Callerid:test Callback Service <4711> The callback is established correctly, but the variable ${CALLERID(num):} is empty The output: NoOp(Calleridnumber: ${CALLERID(num):} is blank. I don.t find my "test Callback Service <4711>" on the cli or as DumpChan(). Under the previous version still has it all works. has something changed? Can you help please? thanks.

Asterisk Users 4.2 years ago 0 Answers

No audio format found to offer.

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This *should* be something that's easy to fix, but apparently I'm not
doing something right. Our SIP long distance provider is telling us to only use formats G.723
and G.729, so I've set up their trunk configuration in sip.conf as such: [t564]
type=friend
host=XXX.XX.56.4
context=default
disallow=all
allow=g723
allow=g729 However, the Dial application gives the following error:

Asterisk Users 4.2 years ago 2 Answers

Asterisk 1.4.42 Now Available (Final Maintenance Release)

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The Asterisk Development Team has announced the final maintenance release of
Asterisk, version 1.4.42. This release is available for immediate
download at
http://downloads.asterisk.org/pub/telephony/asterisk/ Please note that Asterisk 1.4.42 is the final maintenance release from the
1.4 branch. Support for security related issues will continue until
April 21,
2012. For more information about support of the various Asterisk
branches, see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions The release of Asterisk 1.4.42 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you! The following is…

Asterisk Users 4.2 years ago 0 Answers

Asterisk 1.4 func_odbc frustrations

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Maybe somebody can help me here. I've finally got another server together, so I can test and upgrade a
couple of my older 1.4.x installations. I figured that while I'm at it, I'll give func_odbc a try (have been
using the mysql addon), knowing full well that when I finally move over
to 1.8.x, it's what I'm planning on using. I've installed all the requisites listed for ODBC, compiled and install
the current 1.4.41.1 (Was current a couple days ago) and set out
Googling how-tos and digging into voip-info.org After an hour, I had what…

Asterisk Users 4.2 years ago 0 Answers