[Asterisk-Users] Problems with Zpateller on incoming external calls
Title: Problems with Zpateller on incoming external calls 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
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
RE: [Asterisk-Users] Problems with Zpateller on incoming external calls
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