Reid A. Forrest wrote:
Ummm.... X100P IS an analog adaptor.

Well, I thought he was talking about some of the ATA�s available on the market. If an X100P is one analog adapter and has this problem, then everything else has this problem too (except E1/T1/PRI boards that have it�s own signaling).
Ok, so how you guys set up voicemail or anything else that needs more than one priority level? I can�t believe nobody has voicemail on analog lines!! That would be a major flaw in asterisk.
As I and others have said, callprogress is experimental and of not much use. For me it does nothing.


        Gelson


-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gelson Dias Santos Sent: Monday, June 07, 2004 4:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zapata FXO always answers call?

Steven Critchfield wrote:


your using an analog adapter. We cover this regularly. You need
callprogress detection since you don't have a reliable way of doing it
via the analog adapter.


No, I�m not using an analog adapter. As I said, my X100P connects directly to my PBX.
Gelson




On Wed, 2004-06-02 at 14:09, Gelson Dias Santos wrote:


I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule:

exten => _21XX,1,Dial(Zap/g1/${EXTEN:2},20)

where 21 is just a prefix to indicate it�s an analog extension and XX


matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on the log that the call was answered, even if it�s still ringing.
The problem is that now I want to set up voicemail to those analog extensions, but since * says it "answered on first ring" it never goes to the next priority, where voicemail is called.
I tried callprogress=yes on zapata.conf but it has no effect. Here is

a


tipical log from a call I have _not_ answered:

   -- Executing Dial("SIP/2000-a638", "Zap/g1/32|20") in new stack
   -- Called g1/32
   -- Zap/1-1 answered SIP/2000-a638

I�m runnign 0.9.0 from the tar archive. Is this a known bug? Is there

a


workaround for it?

        Gelson
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