Hi Folks,

I just configured a SIP-FRIEND to the card 0709935046 of my test customer. But I tried to registry my SIP phone using:

Authorization User Name: 0709935046 (card number)
Secret: 940632 as (secret field of SIP friend)
Host: 192.168.0.100

But I got this erro message in my SIP phone Registratios error: 404


Do I have to extra configure something to use a2billing with elastix?

Help me please" I lost my day trying to fix it but I couldn't.

it looks,there is no user in elastix site.Please check user on sip.conf under /etc/asterisk

The file sip.conf seems to be managed by freepbx. The a2billing create the file additional_a2billing_sip.conf where a2billing custumer are configured. There aren't references of this file in sip.conf. Should I include this file into sip.conf using directive #include.

My sip.conf:

[general]

; These files will all be included in the [general] context
;
#include sip_general_additional.conf

;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall. For nat'ing you'd need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line [1000](+) in this
;file. Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf

of course you must add this line in to sip.conf (for use sip registration):
#include additional_a2billing_sip.conf

for iax registration the same in to iax.conf
#include additional_a2billing_iax.conf

Ok! Thank you!

hi, i created a a2billing client an all other setup
added line in sip.conf:
#include additional_a2billing_sip.conf

additional sip has the client settings:
[4775407827]
type=friend
username=4775407827
accountcode=4775407827
regexten=4775407827
callerid=757416273042484
amaflags=billing
secret=0287432266
nat=yes
dtmfmode=RFC2833
qualify=yes
canreinvite=yes
disallow=all
allow=ulaw
allow= alaw
allow= gsm
allow= g729
host=dynamic
context=a2billing
regseconds=0
cancallforward=yes

but cant login to a softphone with that client
please can help how to create a sip client in elastic a2billing?
thanks a lot

sorry
i tried again and now it registers ok

problem was when i copy and paste from the 2billing customer screen dont work but when i copy and paste (account-secret) from the conf file it works fine

hi, i got my a2billing customer to log in fine (i use portsip softphone)
now my problem is, i need to do some additional setup so i can dial out?.
i'm getting "called failed request timeout 408"
i see the a2billing extension logged in my CLI> screen sip show peers but i dont get any procces when i dial out.
When i dial out with my freepbx extension configured with my voip provider it dials ok.

thanks a lot

hello everybody, im having the same problem that you had and i add the line #include additional_a2billing_sip.conf but the sip customer continues without registrar

[general]

; These files will all be included in the [general] context
;
#include sip_general_additional.conf

;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going

through a firewall. For nat'ing you'd need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf


;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line [1000](+) in this
;file. Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.

#include additional_a2billing_sip.conf // HERE IS THE INCLUDE
;
#include sip_custom_post.conf

i appreciate your help..thanks