[asterisk-users] How to resolve CallerID from AudioCodes FXO

Mon Feb 26 21:37:25 MST 2007

I have no experience with AudioCodes but it seems that you need to have 
callerID enabled, leave endpoint phone number blank.  Hope this helps.

Maybe some of this info might help:

http://www.voip-info.org/wiki/view/AudioCodes

++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ If caller 
ID is turn on, then freePBX will only record the receiving number.....not 
the line number.
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Well, you 
can fix this, by using the Routing General settings, Audiocodes allows you 
to Prepend the Hunt Group to the number,
You can then use the Manipulation tables, and strip the source number 
(tel-->IP) after Routing.
So if u set each Endpoint up to have a different Hunt Group, you can get it 
to ID the line.

They also have a x-channel header that can be added for you to look in the 
SIP message at.

things that help when dealing with the FXO's

They are designed to work with Analog PSTN lines,
1. Caller ID is usually delivered between the 1st and 2nd ring on these 
lines. Also make sure it is enabled in the Supplementary services.
2. For those of you expecting the number to get delivered through to the IP 
side when dialing, it won't PBX's and CO's just ring the PSTN line, they 
don't deliver the number. Make sure you Enable AutoMatic Dialing in the 
Endpoint Settings, and if you want the line in port x to be the number 
dialed to the sip side, datafill the number there.
3. Make sure you set up the audiocodes with the proper coder like Ulaw, they 
come set to 723 by default which is crap for coders. they can support up to 
5 so just datafill them with all the big coders U, A, 729, and whatever else
4. The Advanced Configuration pages have all their Channel settings, make 
sure the fax's are set to what the Trixbox supports. Audiocodes by default 
does t.38 now. if your pbx isn't set up for it, you need to put the 
Audiocodes in a transparent or events mode
If you want the source  number from IP to use the same datafilled Endpoint 
Port on the PSTN side  make sure  Endpoint Phone Numbers has that number 
datafilled, and then set up a hunt group with source number as the selection 
algorithm(5.0).   Assign the endpoints to that hunt group.   IP to Tel 
rouitng route all calls to that group

Endpoint Phone Number
   - This will give you the options for either 4 or 8 ports.  You do not 
need to place anything here. However, it is a good idea to do such to help 
you identify
      which port the call comes in on; as you can view the reports in 
freePBX to identify calls.  In my case, since I have four PSTN ports, I used 
the last four
      digits of the telephone number to identify.  Identifying which PSTN 
line the call came from only works if you DO NOT have caller id on the line, 
or your
      turn off caller id.  If caller ID is turn on, then freePBX will only 
record the receiving number.....not the line number.
Endpoint Settings
   - Automatic Dialing - Define a station number located on Asterisk / 
Trixbox  (ie 101) for all ports
   - Caller ID - Allowed  ...... turn off if you want to Identify the line 
they came in on.
   - Detect Caller ID from Tel - Enable

Thanks,
Steve Totaro

>From: Angel Heart <cocent at yahoo.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>
>To: jgomez at qualis.com.ar,Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>
>Subject: Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO
>Date: Mon, 26 Feb 2007 19:22:53 -0800 (PST)
>
>Hi      José,
>
>I have not resolve this issue yet. I am currently focusing in my newly 
>arrived toy (fonebridge2) then after which I will go back to AudioCodes 
>Issue.
>
>Still I don't received yet any response from AudioCodes Representative here 
>in the Philippines. I had already escalated this to their Regional Office 
>in Singapore. But still no reply for almost a month already. I will post 
>immediately once I resolve the issue. It is important to us because we 
>really need to now where the calls coming from.
>
>Regards
>
>Angel.
>
>
>
>José Luis Gómez <jgomez at qualis.com.ar> wrote: Hello Angel.
>Did you solve this issue?
>I have the same problem.
>Thanks,
>      José
>
>El jue, 01-02-2007 a las 01:15 -0800, Angel Heart escribió:
> > Hi,
> >
> > I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming
> > & outgoing calls. However, I noticed that the caller ID of the caller
> > coming from the FXO displays its endpoints assigned number and not the
> > actual caller's ID coming from PSTN.
> >
> > Hope someone is using the same scenario and could share on how to
> > resolve the caller ID/Number.
> >
> > Thanks.
> >
> > Angel
> >
> >
> >
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