I bought an Aastra PT480 from digium, but I wanted to see if I could get some more help with this before Monday. Any help would be appreciated. I have the phone connected to the TDM400P card, and I also have the T100P and the X100P in the same box. My problem is, it appears as if the phone and asterisk can't understand each other. The port the phone is connected to always remains off-hook (I hear this is standard for adsi?). The dtmf signals sent to the phone don't ever even break the dialtone coming from asterisk, so I can't dial any extensions. I did then set up the channel to immediate=yes, and set a default extension that waits 3 seconds and tries the ADSIProg application, which from using -vvvvvvvv, it says it returned with a -1, and attempts to hangup the channel, but the phone only receives a fast busy signal. I have also tried this with the GetCPEID application, and that also is returned with a -1. Digium said I should try doing that with the voicemailman application, and I did so, and it did not change anything. The channel does have adsi=yes and rx and txgain set to 0.0. I tried koolstart, loopstart, and groundstart. Kool and loop both did the above, and groundstart, I couldn't get the line to be off-hook. Anyone have any ideas? This is the top of asterisk.adsi: DESCRIPTION "Asterisk PBX" ; Name of vendor VERSION 0x02 ; Version of stuff SECURITY "_AST" ; Security code ;SECURITY 0x0000 ; Security code FDN 0x0000000f ; Descriptor number This is the part of Zapata.conf that is applicable: ;FXS channels via 4 port FXO card usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=no echocancel=yes echocancelwhenbridged=yes relaxdtmf=no rxgain=0.0 txgain=0.0 group=2 callgroup=2 pickupgroup=2 immediate=yes busydetect=no adsi=yes musiconhold=default signalling=fxo_ls context=internal callerid="Internal Unknown"<000> channel => 2-5 -------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20030518/b8bd01f5/attachment.htm
On Sun, May 18, 2003 at 02:54:20AM -0400, Joe Antkowiak wrote:> I bought an Aastra PT480 from digium, but I wanted to see if I could get > some more help with this before Monday. Any help would be appreciated.Well, it's almost Monday here but I'll see if I can help.> I have the phone connected to the TDM400P card, and I also have the T100P > and the X100P in the same box.That sounds about right.> My problem is, it appears as if the phone and asterisk can't understand each > other. The port the phone is connected to always remains off-hook (I hear > this is standard for adsi?).No. The phone should be onhook unless it's pioked up. I think this may be an indication that your signalling is not set up correctly.> The dtmf signals sent to the phone don't ever > even break the dialtone coming from asterisk, so I can't dial any > extensions. I did then set up the channel to immediate=yes, and set a > default extension that waits 3 seconds and tries the ADSIProg application, > which from using -vvvvvvvv, it says it returned with a -1, and attempts to > hangup the channel, but the phone only receives a fast busy signal. I have > also tried this with the GetCPEID application, and that also is returned > with a -1. Digium said I should try doing that with the voicemailman > application, and I did so, and it did not change anything.This does not sound at all normal. You definitely shouldn't need immediate=yes.> The channel does have adsi=yes and rx and txgain set to 0.0. I tried > koolstart, loopstart, and groundstart. Kool and loop both did the above, > and groundstart, I couldn't get the line to be off-hook.Are you trying fxs or fxo? I think your zapata.conf should be using something like fxo (even though they're fxs lines, you're signalling to fxo devices). Is that right? Somebody may pipe in here because this last bit has always confused me.> Anyone have any ideas?See about, good luck.> SECURITY "_AST" ; Security code > FDN 0x0000000f ; Descriptor numberThese two values may be problematic for you. Yhey are specific to the type (not just he model number, but actually what region/vendor they were intended for). You may not be able to get the number for yours, although I can set you up with the right ones. Now, right now, it sounds like your phones and your card aren't talking--but when they are, this will be your next big problem.> This is the part of Zapata.conf that is applicable: > signalling=fxo_lsThis looks right.> channel => 2-5This would be right only if the X100P is being detected first, then the TDM400P and then the T100P. MAKE SURE THIS IS THE CASE! Otherwise, you may be signalling the wrong channel. If you haven't used it, zttool might give some hint here. Otherwise, I'm not really sure how to tell (ztcfg, anyone?). Jayson
>> My problem is, it appears as if the phone and asterisk can't understandeach>> other. The port the phone is connected to always remains off-hook (Ihear>> this is standard for adsi?). >No. The phone should be onhook unless it's pioked up. I think this may >be an indication that your signalling is not set up correctly.>Actually, if the lights are on all the time, it means the phone is NOT >connected to the line !!The phone *is* connected to the zaptel card, and I am getting a dialtone, so I'm assuming my problem is the code/fdn I have set in asterisk.adsi...
>> SECURITY "_AST" ; Security code >> FDN 0x0000000f ; Descriptor number >These two values may be problematic for you. Yhey are specific to the >type (not just he model number, but actually what region/vendor they >were intended for). You may not be able to get the number for yours, >although I can set you up with the right ones. Now, right now, it >sounds like your phones and your card aren't talking--but when they are, >this will be your next big problem.I'm thinking this might be the whole problem... I'll be trying out some different codes that someone was nice enough to send me...>> This is the part of Zapata.conf that is applicable: >> signalling=fxo_ls >This looks right.Yep.>> channel => 2-5 >This would be right only if the X100P is being detected first, then the >TDM400P and then the T100P. MAKE SURE THIS IS THE CASE! Otherwise, you >may be signalling the wrong channel. If you haven't used it, zttool >might give some hint here. Otherwise, I'm not really sure how to tell >(ztcfg, anyone?).I do have my channels configured properly...
A normal tone phone *does* work on that same port (way ahead of you), and the adsi phone works fine on a normal telco POTS line (minus the features) -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Gary Sent: Monday, May 19, 2003 8:54 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] little ADSI problem forget the fact is an ADSI phone, if it wont work like a normal phone it is probably stuffed. Start with the basics and get that working, then work up... got a normal tone phone, does it work properly on that port ? then continue etc..... On Mon, 19 May 2003 08:43:16 -0400, Joe Antkowiak wrote:>>> My problem is, it appears as if the phone and asterisk can't understand >each >>> other. The port the phone is connected to always remains off-hook (I >hear >>> this is standard for adsi?). >>No. The phone should be onhook unless it's pioked up. I think this may >>be an indication that your signalling is not set up correctly. > > >>Actually, if the lights are on all the time, it means the phone is NOT >>connected to the line !! > >The phone *is* connected to the zaptel card, and I am getting a dialtone,so>I'm assuming my problem is the code/fdn I have set in asterisk.adsi... > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >lists.digium.com/mailman/listinfo/asterisk-users. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users