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

Mon Feb 26 21:45:45 MST 2007

This sounds promising and obvious.  Two rings is pretty standard for analog 
caller ID.

I would like to add some additional insight into #7. Unless you have the 
MP-114 hanging off a PBX with caller ID, you should probably set the 
PROTOCOL MANAGEMENT > FXO SETTINGS > Rings before detecting caller ID to 1. 
Most Class 5 offices will not give you caller ID messaging info until 
sometime during the first cycle; as this info comes from the SS7 channel 
with lower priority than the actual signaling info. Whereas, if you are 
hanging off a PBX, by the time the PBX trunk recognizes a seizure, the 
caller ID info is delivered and the PBX makes cut-through to the station 
delivering both ring generator and caller ID as the same time. Setting it to 
0 off a class 5 could either give you the caller ID content you defined in 
the station ID info in ENDPOINT fields, or if you put nothing in there, 
unknown caller, or 1000 (default setting PROTOCOL MANAGEMENT > PROTOCOL 
DEFINITION > DTMF & DIALING as you defined in #5.

>From: "Steven Totaro" <stevetotaro at hotmail.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>
>To: asterisk-users at lists.digium.com
>Subject: Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO
>Date: Mon, 26 Feb 2007 23:37:25 -0500
>
>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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