Tried that, and no go. There's something wrong with Zapteller. It works fine
on internal calls, but the only way I can get it to work on external calls
(through a SIP/PSTN gateway, no Zap hw necessary) is to first play a
message. For instance, this works:
exten = 2200,1,Playback(ss-noservice)
exten = 2200,2,Zapateller
exten = 2200,3,Dial(SIP/2205)
-brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Thompson
Sent: Friday, April 09, 2004 12:48 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Problems with Zpateller on
incoming external calls
Brian Cuthie wrote:
I've setup the following in extensions.con:
exten = 2200,1,Ringing
exten = 2200,2,Wait(2)
exten = 2200,3,Answer
exten = 2200,4,Zapateller
exten = 2200,5,Macro(stdexten,2205,SIP/2205)
This works as expected if I dial from a SIP phone on my desk.
However, if I dial in from the PSTN (through a SIP
provider) it fails
while trying to play ths SIT with: Apr 8 18:53:12
WARNING[1209269552]: rtp.c:407 ast_rtp_read: RTP Read
error: Resource
temporarily unavailable
Any idea what's going on? My suspicion is that the PSTN gateway
hasn't setup an audio path yet, although I thought Answer would do
that.
Cheers,
Brian
I don't have a zap device to test on, but can you do Ringing
before you Answer?
-
Andrew Thompson
http://aktzero.com/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
