[asterisk-users] SIP 603 Declined after AGI execution

2011-05-19 Thread Alejandro Mejia Evertsz 

Hello everyone.

I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small 
wholesale operation, so I configured A2Billing for not to answer the 
call nor play any greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer 
has it's own context, in which I set the following:


;=in extensions.conf==
[from-customerX]
exten = _X.,1,Set(CDR(accountcode)=)  ;Here I change the 
accountcode depending on each customer

exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
exten = _X.,3,Goto(a2billing|${EXTEN}|1)

[from-customerY]
exten = _X.,1,Set(CDR(accountcode)=)
exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
exten = _X.,3,Goto(a2billing|${EXTEN}|1)

[a2billing]
exten = _X.,1,DeadAGI(a2billing.php|1)
exten = _X.,2,Hangup(34)
;=

A2Billing authenticates and routes the call properly, but when the 
termination gateway for the destination dialed by the customer rejects 
the call, my Asterisk box sends 603 Declined to the customer.
It also happens when A2Billing doesn't find any route for that 
destination, in which it should return 404 Not Found, but returns 603 
Declined instead.
I tried to force every rejected attempt with 503 Service Unavailable 
putting the Hangup(34) you see on my config, but it never seems to get 
there.
The last thing I see on CLI running in verbose is: -- AGI Script 
a2billing.php completed, returning 0


Is there anything I could do to return a different cause than 603 
Declined?

I posted the same question on A2Billing's forum, but had no luck.

Thanks in advance,

Alejandro Mejia

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[asterisk-users] Getting 603 Declined after AGI execution

2011-05-19 Thread Alejandro Mejia Evertsz 

Hello everyone.

I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small 
wholesale operation, so I configured A2Billing for not to answer the 
call nor play any greetings or balance notifications to the caller.
I'm authenticating each customer by it's IP address, and each customer 
has it's own context, in which I set the following:


;=in extensions.conf==
[from-customerX]
exten = _X.,1,Set(CDR(accountcode)=)  ;Here I change the 
accountcode depending on each customer

exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
exten = _X.,3,Goto(a2billing|${EXTEN}|1)

[from-customerY]
exten = _X.,1,Set(CDR(accountcode)=)
exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
exten = _X.,3,Goto(a2billing|${EXTEN}|1)

[a2billing]
exten = _X.,1,DeadAGI(a2billing.php|1)
exten = _X.,2,Hangup(34)
;=

A2Billing authenticates and routes the call properly, but when the 
termination gateway for the destination dialed by the customer rejects 
the call, my Asterisk box sends 603 Declined to the customer.
It also happens when A2Billing doesn't find any route for that 
destination, in which it should return 404 Not Found, but returns 603 
Declined instead.
I tried to force every rejected attempt with 503 Service Unavailable 
putting the Hangup(34) you see on my config, but it never seems to get 
there.
The last thing I see on CLI running in verbose is: -- AGI Script 
a2billing.php completed, returning 0


Is there anything I could do to return a different cause than 603 
Declined?

I posted the same question on A2Billing's forum, but had no luck.

Thanks in advance,

Alejandro Mejia

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Re: [asterisk-users] click to call with php

2011-05-19 Thread Alejandro Mejia Evertsz 
You only need to tell your PHP script to write a .call file on 
/var/spool/asterisk/outgoing/ directory using the syntax described here:

http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out

I'm not a PHP programmer, so the PHP part is up to you hehe.

There are other methods like using manager, but to keep it simple, I 
recommend you to use .call files.


Good luck...

On 19/05/2011 10:44 a.m., salaheddine elharit wrote:


Hello,

i have asterisk 1.4 installed and i want to use click to call in order 
to do an outbound call


if there is any php code in order to do this operation

thanks and regards


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[Asterisk-Users] cdr_addon_mysql can't find libmysqlclient.so

2005-12-13 Thread Alejandro Mejia Evertsz 



Hello 
list!
I had a problem 
while trying to build asterisk-addons, but noticed some paths specified in the 
Makefile didn't fit my system.
So I modified 
Makefile for it to look for MySQL includes and libs on the following 
locations:

/usr/local/mysql/include/mysql
/usr/local/mysql/lib/mysql

Now when trying 
"make" it works fine, and "make install" too. ;)
But when I add 
cdr_addon_mysql.so on modules.conf for Asterisk to load it, Asterisk refuses tu 
come up saying:

Dec 13 12:19:29 
WARNING[4112]: loader.c:325 __load_resource: libmysqlclient.so.15: cannot open 
shared object file: No such file or directoryDec 13 12:19:29 WARNING[4112]: 
loader.c:499 load_modules: Loading module cdr_addon_mysql.so 
failed!
Is there any other 
place in which I should specify the diffetent locations for my 
system?
How can I fix 
this?

Thanks for your 
help.

Cheers!
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RE: [Asterisk-Users] call levels

2005-11-17 Thread Alejandro Mejia Evertsz 



Hi,

One solution would be to configureeach IP phone 
(in sip.conf) on a different context.
[phone1]
context=local

[phone2]
context=intl

Then make 2 different contexts named "local"and 
"intl" on extensions.conf

[local]
exten = _9X.,1,yourDialString
; Nine followed by any number is just an example as I 
don't know the dial plan in Argentina

[intl]
include = local
; Including local context to be available on intl 
too
exten = _00X.,1,yourDialString


This just looks like crap if you hadn't take the time 
to read a bit about the dialplan.
I recommend you to look at:

http://www.voip-info.org/wiki-Asterisk+config+extensions.conf

voip-info.org has lots of nice documentation 
;)

Hope it helps a bit...


Buena suerte!

Alejandro Mejia


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Mariano 
GonzalezSent: Jueves, 17 de Noviembre de 2005 03:52 
p.m.To: asterisk-users@lists.digium.comSubject: 
[Asterisk-Users] call levels


Hello all.
This is my first time with Asterisk, may be my 
question is fool.
I have a two IP phone.
I need that the first phone makes calls to local 
numbers only and the second phone make calls to all numbers.
Somebody know the solution?
Thanks a lot.

Mariano

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[Asterisk-Users] TDM04b on FreeBSD

2005-11-16 Thread Alejandro Mejia Evertsz 



Hi 
list!

I successfully 
installed a Digium TDM04B card on FreeBSD 5.4 using zaptel drivers for FreeBSD 
(installed with ports).
I'm using Asterisk 
CVS-Head and the card works fine, but when placing or recieving a call on any of 
the 4 fxo ports, users hear (both sides) a "clicking" noise.
I also have a 
Wildcard X100P installed, and uses the same configuration (on zapata.conf) but 
that card doesn't make that strange noise during 
conversations.

Bellow, my 
zapata.conf (in case I'm doing something wrong) so you can suggest any changes 
or correct my mistakes.

===zapata.conf
[channels];context=defaultsignalling=fxs_ksusecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yes
echocancel=yesechocancelwhenbridged=yes
rxgain=0.0txgain=0.0
immediate=yes
busydetect=yesbusycount=4

jitterbuffers=4

group=0
channel = 
1
group=1channel 
= 2
group=2channel 
= 3

group=3channel 
= 4

group=4channel 
= 5

==END OF 
FILE=

Explanation:
Channel 1 = 
Wildcard
Channels 2-5 = 
TDM04b

Please let me know 
if someone had this problem before me, and what you did to correct 
it.
I don't know what 
else to try.

Thank you 
all.


Alejandro 
Mejia
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[Asterisk-Users] 407 Proxy Authentication Required

2005-02-25 Thread Alejandro Mejia Evertsz 
Hi everybody:

I configured my Asterisk to register to my VoIP provider, and I can make
outgoing calls, but I can't receive any calls with it.
I used Ethereal to sniff the activity of it, and I found something that
might be causing the problem:
When my provider's gateway does the Request: INVITE
[EMAIL PROTECTED] ... my Asterisk asks for Status: 407 Proxy
Authentication Required, (log line 10) but my provider's gateway never
sends this info back, so my Asterisk keeps on asking for the Authentication,
and it never comes back... so it gives a time-out (I guess).

What I need to know is how to configure my Asterisk for not to ask for
Authentication.

Here's the log if you would like to see what's going on:
192.168.1.116 = ATA from which I'm calling [EMAIL PROTECTED]
192.168.1.48  = My Asterisk server

Thank you ;)

No. TimeSourceDestination   Protocol
Info
  1 0.00192.168.1.116 VoIP Prov IP  SIP/SDP
Request: INVITE sip:[EMAIL PROTECTED], with session description
  2 0.369430VoIP Prov IP  192.168.1.116 SIP
Status: 100 Trying
  3 0.401052VoIP Prov IP  192.168.1.116 SIP
Status: 407 Proxy Authentication Required
  4 0.407666192.168.1.116 VoIP Prov IP  SIP
Request: ACK sip:[EMAIL PROTECTED]
  5 0.414146192.168.1.116 VoIP Prov IP  SIP/SDP
Request: INVITE sip:[EMAIL PROTECTED], with session description
  6 0.907932192.168.1.116 VoIP Prov IP  SIP/SDP
Request: INVITE sip:[EMAIL PROTECTED], with session description
  7 1.541468VoIP Prov IP  192.168.1.116 SIP
Status: 100 Trying
  8 1.563302VoIP Prov IP  192.168.1.116 SIP
Status: 180 Ringing
  9 1.635021VoIP Prov IP  192.168.1.48  SIP/SDP
Request: INVITE sip:[EMAIL PROTECTED]:5060;maddr=192.168.1.48, with
session description
 10 1.636719192.168.1.48  VoIP Prov IP  SIP
Status: 407 Proxy Authentication Required
 11 1.653490VoIP Prov IP  192.168.1.116 SIP
Status: 100 Trying
 12 1.686395VoIP Prov IP  192.168.1.48  SIP
Request: OPTIONS sip:[EMAIL PROTECTED]:5061
 13 2.637223192.168.1.48  VoIP Prov IP  SIP
Status: 407 Proxy Authentication Required
 14 3.647291192.168.1.48  VoIP Prov IP  SIP
Status: 407 Proxy Authentication Required
 15 3.887926VoIP Prov IP  192.168.1.116 SIP
Request: OPTIONS sip:[EMAIL PROTECTED]
 16 3.897185192.168.1.116 VoIP Prov IP  SIP
Status: 200 OK
 17 4.119698VoIP Prov IP  192.168.1.48  SIP
Request: OPTIONS sip:[EMAIL PROTECTED]
 18 4.120788192.168.1.48  VoIP Prov IP  SIP
Status: 200 OK
 19 4.647336192.168.1.48  VoIP Prov IP  SIP
Status: 407 Proxy Authentication Required
 20 5.647409192.168.1.48  VoIP Prov IP  SIP
Status: 407 Proxy Authentication Required
 21 6.647465192.168.1.48  VoIP Prov IP  SIP
Status: 407 Proxy Authentication Required
 22 7.657954VoIP Prov IP  192.168.1.116 SIP
Status: 180 Ringing

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