Voxilla VoIP Forum - Asterisk Support Forum

Asterisk Programming

husainsn — Sat, 04 Jul 2009 02:07:57 GMT

Hello Bobzee

I am trying to run an agi application which comes with Asterisk. I inserted the following dial plan in “extensions-custom.conf” file:

[custom-snhagi]
exten => 300,1,Answer
exten => 300,n,AGI(agi-test.agi)
exten => 300,n,Hangup

My newbie question is how do I invoke this dial plan? I created an extension 300 and placed it in a ring group. When extension 300 is answered I receive silence then prompt from voice mail for unanswered call.

Thanks

Function ODBC_ not registered

dermosko — Thu, 02 Jul 2009 20:12:10 GMT

Hi,

Running 1.6, when callin g ODBC_function from dialplan , I'm getting error Function ODBC_funcname not registered. I thought that maybe I left it out during compilation, but don't see such option in "makemenueselect".

thanks

Asterisk Syntax

pmoore4321 — Thu, 02 Jul 2009 17:51:25 GMT

freepbx

Trunk Name: ipkall
PEER Details:

type=peer
qualify=yes
host=voiper.ipkall.com
disallow=all
context=from-pstn
allow=ulaw

Everything else is blank.

source:
http://samyantoun.50webs.com/asteris.../setup/ipkall/

Add jabber

jonaskellens — Wed, 01 Jul 2009 07:50:19 GMT

Quote:

Originally Posted by wtei (Post 129608)

I guess my question is how do I install this new feature that is not installed/disabled without re-installing asterisk since I am on a live system?

The module res_jabber depends on iksemel and gnutls. You need to have these installed on your Centos-system before the module becomes available in menuselect.
You then need to recompile your Asterisk.
You can do a make clean, make. Make install will put all the files in the correct directory and to my guess will overwrite them. After you ran make, you have the module res_jabber.so in your current directory where you ran the make-command. You can copy this module to the modules-directory of Asterisk.
No existing files will be overwritten, no live system will be affected...
Don't do a make install, just 'make clean' and 'make'. And then manual copy of the module.

After that, the jabber-module can be loaded in modules.conf, and you can utilize commands like

Code:

exten => 10,1,JABBERSend(asterisk,user@host.tld,

Call from"${CALLERID(name)}" at number <${CALLERID(num)}>

on ${STRFTIME(,EST5EDT,%A %B %d %G at%l:%M:%S %p)} )

Asterisk: no outgoing audio

jonaskellens — Wed, 01 Jul 2009 07:43:18 GMT

I have changed my setup. Asterisk has now just 1 interface on the orange network (DMZ) and not as before 2 interfaces (so there is no more routing on the Asterisk-server).

My phones register to the orange IP-address. The firewall is in between.

Registration succeeds and there is audio in both ways with the following SIP-config :

[3starsnet]
type=peer
host=ip_provider
username=my_user
secret=XXXX
fromuser=my_user
fromdomain=sip.3starsnet.com
dtmfmode=rfc2833
canreinvite=no
insecure=port,invite
qualify=yes
nat=yes
disallow=all
allow=gsm
allow=alaw

I do not have 'externip' and 'localnet' defined in [general]...

Thanks for the help !

Click to call, How to get Status of call

bobzee — Tue, 30 Jun 2009 18:10:52 GMT

I am trying to implement a web based click to call and using AMI to Originate call to both parties.... Now if I want to show status of call like "Ringing", "call in progress" or "Hangup" etc.... how can I can do it.... I tried to use "Status" or "ExtensionState" actions without much luck.

Any help is appreciated

Bob

FreePBX in a cloud - FreePBX secured and optimized for Amazon EC2

ChrisDC — Mon, 29 Jun 2009 04:18:39 GMT

I have a newbie question-- what is the best IAX/SIP trunk to connect to Asterisk in EC2? I have an application that makes calls across the US only. Thanks in advance for any guidance on a good provider!

Asterisk and Billing software

qwerty76 — Sun, 28 Jun 2009 17:48:21 GMT

Hi,

i am looking for an Billing software. But I am running Asterisk v 1.4

I saw some software for the v 1.2, but nothing for 1.4
is there anybody, who can tell me which Software I can use for Asterisk Version 1,4?

Thanks

SPA400 SIP/2.0 489 Bad event

oconmx — Tue, 23 Jun 2009 20:22:57 GMT

Any idea? not even a hint :(

Thanks!

Asterisk 1.4 + mISDN + HFC-PCI on Bosch/Tenovis Integral 3 PBX possible?

adiadi — Mon, 22 Jun 2009 06:24:02 GMT

Hi,
on Tenovis PBX for the respective SO in prog.41 set 1, 2, 3. That setting is for ISDN terminal equipment (all except tenovis) incomming calls.
Hope this helps.

EZ-install Shorewall Firewall for FreePBX

pmoore4321 — Sun, 14 Jun 2009 03:57:51 GMT

This script will install the Shorewall firewall on an Asterisk/Freepbx system running Centos. This is for 32-bit machines, 64-bit machines must make appropriate changes.

http://www.prestonmoore.com/images/c...bx-firewall.sh

Code:

# Install shorewall firewall

cd /usr/src/

wget http://www.invoca.ch/pub/packages/shorewall/4.2/shorewall-4.2.9/shorewall-4.2.9-1.noarch.rpm

wget http://www.invoca.ch/pub/packages/shorewall/4.2/shorewall-4.2.9/shorewall-perl-4.2.9-1.noarch.rpm

wget http://www.invoca.ch/pub/packages/shorewall/4.2/shorewall-4.2.9/shorewall-shell-4.2.9-1.noarch.rpm

rpm -ivh shorewall-perl-4.2.9-1.noarch.rpm shorewall-shell-4.2.9-1.noarch.rpm shorewall-4.2.9-1.noarch.rpm



# Install ulogd netfilter logging program

wget ftp://ftp.pbone.net/mirror/rpm.razorsedge.org/centos-5/RE-test/ulogd-1.24-2.el5.re.i386.rpm

rpm -ivh ulogd-1.24-2.el5.re.i386.rpm

sed -i "s/loglevel=5/loglevel=3/" /etc/ulogd.conf



# Configure shorewall for asterisk / freepbx

cd /etc/shorewall/

mkdir shorewall-default

mv * /etc/shorewall/shorewall-default/

wget http://www.prestonmoore.com/images/shorewall-asterisk.tar.gz

tar xvfz shorewall-asterisk.tar.gz

cp -v /etc/shorewall/ast-macros/* /usr/share/shorewall/

chkconfig ulogd on

chkconfig shorewall on

/etc/init.d/ulogd start

/etc/init.d/shorewall start

echo

echo "The following firewall ports are open by default:"

echo

echo "    SSH        22/tcp"

echo "    HTTP        80/tcp"

echo "    Flash Panel       4445/tcp"

echo "    IAX        4569/udp"

echo "    SIP        5060/udp"

echo "    RTP           10000:20000/udp"

echo

echo "Firewall rules are set in /etc/shorewall/rules"

echo

Asterisk FastAGI

Sara Ibn El Ahrache — Thu, 11 Jun 2009 10:45:00 GMT

I am writing multiple agis using perl that will be called from the asterisk dialplan. I expect to receive numerous similtaneous calls so I need a way to load balance them. I have been advised to use fastagi instead of agi. The problem is that my agis will be distributed over many servers not just one, and I need that my entry point asterisk dispatches the calls among those servers (where the agis reside) based on their availability. So, I thought of providing the fastagi application with multiple ip addresses instead of one, is it possible?

Howto: Multiple registrations from same provider?

medisoft — Wed, 10 Jun 2009 23:21:31 GMT

Hi! I'm trying to config my asterisk for multiple registrations from same provider.

I have 2 numbers from same service, "serviceip.com", with username "aaaa" and "bbbb" and password "xxxx" and "yyyy" and phone number "1234-5678" and "1234-6789"

i have two context "[contexta]" and "[contextb]"

i also have multiple extensions in both contexts.

i wish that when someone calls to "1234-5678" the call be redirected to extension "s" on "[contexta]" and when someone calls to "1234-6789" the call be redirected to extension "s" on "[contextb]"

also, when some extension in "[contexta]" dials a number (like 8243-2732) the call uses the right registry (user aaaa)

right now i can do this with only one registry, but when i activate both of them, then nothing works, no in calls, no out calls, only calls between extensions of the same context.

in sip.conf i have this:

register => aaaa:xxxx@serviceip.com/99900
register => bbbb:yyyy@serviceip.com/99901

[myphone]
context=contexta
type=friend
host=dynamic
mailbox=myextensionnumber
username=myphone
secret=****
qualify=yes
callerid="My Name"
nat=yes
canreinvite=yes

[myotherphone]
context=contextb
type=friend
host=dynamic
mailbox=myotherextensionnumber
username=myotherphone
secret=****
qualify=yes
callerid="My Name"
nat=yes
canreinvite=yes

and on extensions.conf

[default]
exten => 99900,1,Goto(contexta,s,1)
exten => 99901,1,Goto(contextb,s,1)

on [contexta] and [contextb] i have the "demo" context that comes with asterisk, only copied it and changed the context name to contexta and contextb

Asterisk / Freepbx Centos 5.3

pmoore4321 — Wed, 10 Jun 2009 07:09:59 GMT

http://www.prestonmoore.com/archives...entos-5.3.html

Trunk Sharing Forums have arrived!

markosjal — Tue, 09 Jun 2009 22:32:08 GMT

Just an FYI we are currently offering new forums that allow user to user trunk sharing. The idea is that users that pay flat rate to a destination can sell off a part of their available capacity whether 1 or 100 channels. This is not for wholesale termination , nor is it for terminations sold by the minute.

It represents a chance for those with extra capacity to help someone else while both may keep a little more in their pocket. We want to move VoIP from wall street to main street.

We have a very reasonable fee and will in fact waive the fee to some new listers who have good capacity for a good price.

Please visit Teknogeekz.com
or
http://www.teknogeekz.com/trunkforums.htm for the info page.