Hi Steven,<br><br>Thank you for your response. I had tried leaving endpoint phone number blank but when I tried making an outside call, the Audiocodes seems doen't know where to pass the call. So I need to assigned numbers. There is no problem for incoming call aside from not being displayed the Caller ID.<br><br><br><br><b><i>Steven Totaro <stevetotaro@hotmail.com></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> I have no experience with AudioCodes but it seems that you need to have <br>callerID enabled, leave endpoint phone number blank. Hope this helps.<br><br>Maybe some of this info might help:<br><br>http://www.voip-info.org/wiki/view/AudioCodes<br><br>++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ If caller <br>ID is turn on, then freePBX will only record the receiving number.....not <br>the line
number.<br>++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Well, you <br>can fix this, by using the Routing General settings, Audiocodes allows you <br>to Prepend the Hunt Group to the number,<br>You can then use the Manipulation tables, and strip the source number <br>(tel-->IP) after Routing.<br>So if u set each Endpoint up to have a different Hunt Group, you can get it <br>to ID the line.<br><br>They also have a x-channel header that can be added for you to look in the <br>SIP message at.<br><br>things that help when dealing with the FXO's<br><br>They are designed to work with Analog PSTN lines,<br>1. Caller ID is usually delivered between the 1st and 2nd ring on these <br>lines. Also make sure it is enabled in the Supplementary services.<br>2. For those of you expecting the number to get delivered through to the IP <br>side when dialing, it won't PBX's and CO's just ring the PSTN line, they <br>don't deliver the number. Make sure you Enable
AutoMatic Dialing in the <br>Endpoint Settings, and if you want the line in port x to be the number <br>dialed to the sip side, datafill the number there.<br>3. Make sure you set up the audiocodes with the proper coder like Ulaw, they <br>come set to 723 by default which is crap for coders. they can support up to <br>5 so just datafill them with all the big coders U, A, 729, and whatever else<br>4. The Advanced Configuration pages have all their Channel settings, make <br>sure the fax's are set to what the Trixbox supports. Audiocodes by default <br>does t.38 now. if your pbx isn't set up for it, you need to put the <br>Audiocodes in a transparent or events mode<br>If you want the source number from IP to use the same datafilled Endpoint <br>Port on the PSTN side make sure Endpoint Phone Numbers has that number <br>datafilled, and then set up a hunt group with source number as the selection <br>algorithm(5.0). Assign the endpoints to that hunt group. IP to Tel
<br>rouitng route all calls to that group<br><br>Endpoint Phone Number<br> - This will give you the options for either 4 or 8 ports. You do not <br>need to place anything here. However, it is a good idea to do such to help <br>you identify<br> which port the call comes in on; as you can view the reports in <br>freePBX to identify calls. In my case, since I have four PSTN ports, I used <br>the last four<br> digits of the telephone number to identify. Identifying which PSTN <br>line the call came from only works if you DO NOT have caller id on the line, <br>or your<br> turn off caller id. If caller ID is turn on, then freePBX will only <br>record the receiving number.....not the line number.<br>Endpoint Settings<br> - Automatic Dialing - Define a station number located on Asterisk / <br>Trixbox (ie 101) for all ports<br> - Caller ID - Allowed ...... turn off if you want to Identify the line <br>they came in on.<br> - Detect Caller ID from Tel -
Enable<br><br>Thanks,<br>Steve Totaro<br><br><br>>From: Angel Heart <cocent@yahoo.com><br>>Reply-To: Asterisk Users Mailing List - Non-Commercial <br>>Discussion<asterisk-users@lists.digium.com><br>>To: jgomez@qualis.com.ar,Asterisk Users Mailing List - Non-Commercial <br>>Discussion<asterisk-users@lists.digium.com><br>>Subject: Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO<br>>Date: Mon, 26 Feb 2007 19:22:53 -0800 (PST)<br>><br>>Hi José,<br>><br>>I have not resolve this issue yet. I am currently focusing in my newly <br>>arrived toy (fonebridge2) then after which I will go back to AudioCodes <br>>Issue.<br>><br>>Still I don't received yet any response from AudioCodes Representative here <br>>in the Philippines. I had already escalated this to their Regional Office <br>>in Singapore. But still no reply for almost a month already. I will post <br>>immediately once I resolve the issue. It is
important to us because we <br>>really need to now where the calls coming from.<br>><br>>Regards<br>><br>>Angel.<br>><br>><br>><br>>José Luis Gómez <jgomez@qualis.com.ar> wrote: Hello Angel.<br>>Did you solve this issue?<br>>I have the same problem.<br>>Thanks,<br>> José<br>><br>>El jue, 01-02-2007 a las 01:15 -0800, Angel Heart escribió:<br>> > Hi,<br>> ><br>> > I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming<br>> > & outgoing calls. However, I noticed that the caller ID of the caller<br>> > coming from the FXO displays its endpoints assigned number and not the<br>> > actual caller's ID coming from PSTN.<br>> ><br>> > Hope someone is using the same scenario and could share on how to<br>> > resolve the caller ID/Number.<br>> ><br>> > Thanks.<br>> ><br>> > Angel<br>> ><br>> ><br>> ><br>> >
______________________________________________________________________<br>> > Bored stiff? Loosen up...<br>> > Download and play hundreds of games for free on Yahoo! Games.<br>> > _______________________________________________<br>> > --Bandwidth and Colocation provided by Easynews.com --<br>> ><br>> > asterisk-users mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > http://lists.digium.com/mailman/listinfo/asterisk-users<br>><br>>_______________________________________________<br>>--Bandwidth and Colocation provided by Easynews.com --<br>><br>>asterisk-users mailing list<br>>To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>><br>><br>><br>>---------------------------------<br>>Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and <br>>get things done
faster.<br><br><br>>_______________________________________________<br>>--Bandwidth and Colocation provided by Easynews.com --<br>><br>>asterisk-users mailing list<br>>To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br><br>_________________________________________________________________<br>Win a Zune™—make MSN® your homepage for your chance to win! <br>http://homepage.msn.com/zune?icid=hmetagline<br><br>_______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br></jgomez@qualis.com.ar></asterisk-users@lists.digium.com></asterisk-users@lists.digium.com></cocent@yahoo.com></blockquote><br><p> 
<hr size=1>8:00? 8:25? 8:40? <a href="
http://tools.search.yahoo.com/shortcuts/?fr=oni_on_mail&#news"> Find a flick</a> in no time<br> with the<a href="
http://tools.search.yahoo.com/shortcuts/?fr=oni_on_mail&#news">Yahoo! Search movie showtime shortcut.</a>