i think i have similar problem after upgraded from 1.4.x to 1.6.2.17.
(originally upgraded to 1.8.3.2 unfortunately there were other more
pressing problems that forced me to downgraded it to 1.6.2.17)
i have a wanpipe device with 2 channels uses PRI signalling to PSTN
the other 2 uses FXO
I had problems with a system I was trying to bring up using a couple older
a104d cards we had lying around. Neither card would pass audio. I worked with
one Sangoma tech for a couple hours while he tried various things. The second
tech I worked with got on the system and updated the firmware
I did more testing.
Here is a portion of extensions.conf on asterisk-pri:
exten = 5,1,Dial(DAHDI/g1/14186939930,30)
exten = 6,1,Answer
exten = 6,2,Wait(30)
exten = 7,1,Dial(DAHDI/g1/14186939930,30,D(132412983#))
Here is an expert from asterisk :
exten = 22,1,Dial(SIP/6@pri,30,D(132412983#))
Hello,
I traced the SIP packets and saw that the only difference was that the
DAHDI channel returns 183 Session progress ( besides the obvious
differences such as the To and from tags in sip , session id and rtp
ports in the SDP ).
I updated my dialplan on asterisk-pri as follows :
exten =
Hello,
I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple
problems with DTMF.
I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does
IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two
servers communicate via SIP with
