HI,
I'm using asterisk + voicetronix openswitch12 (using fxo). I just
noticed when I call a pstn number (mobile number), asterisk will
answer the call first before it actually dials the destination
number. Is this normal?
-- Executing Wait("SIP/192.168.1.130-081671b0", "1") in new stack
-- Executing Dial("SIP/192.168.1.130-081671b0",
"vpb/1-12/9111111") in new stack
== 1-12 requested, got: [vpb/1-12]
== vpb/1-12: Calling 9111111 on vpb/1-12
== vpb/1-12: Dial parms for vpb/1-12 1/2000ms/4000ms/4000ms/120000ms
== vpb/1-12: Dial parms for vpb/1-12 tone 7->0
== vpb/1-12: Dial parms for vpb/1-12 tone 0->1
== vpb/1-12: Dial parms for vpb/1-12 tone 4->2
== vpb/1-12: Dial parms for vpb/1-12 tone 7->3
== vpb/1-12: Dial parms for vpb/1-12 tone 3->4
-- vpb/1-12: VPB Calling 9111111 [t=120000] on vpb/1-12 returned 0
vpb/1-12: chanreads: starting thread
-- Called 1-12/9111111
-- vpb/1-12 is ringing
== vpb/1-12: Dialend
-- vpb/1-12 answered SIP/192.168.1.130-081671b0
== vpb/1-12:Now listening for DTMF
== vpb/1-12: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR]
vpb/1-12: vpb_write: Starting play mode
(codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR]
== vpb/1-12: Hangup requested
== vpb/1-12: Ending record mode (1/yes)
== vpb/1-12: Ending play mode on vpb/1-12
Regards,
Antonio
_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.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
