[asterisk-dev] Asterisk 1.6 - Reading DTMF from AGI

[Ralfe Poisson]

Hi,

I'm busy developing and AGI application which was working well under 1.4 .
I've just upgraded to Asterisk 1.6 and am having two issues which I can't
seem to sort out.

*1. Reading DTMF tones
*
I used to use the following code to play an ivr file and read in dtmf:

#php
fwrite(STDOUT, "EXEC Read wc-ivr|$file|$max_digits\n");
fflush(STDOUT);
fwrite(STDOUT, "GET VARIABLE wc-ivr\n");
fflush(STDOUT);
//handle reply here....

That worked fine, but now, this command seems to just hang. I'm not too sure
what to do. I've tried using the GET DATA command, but that tells me that it
is playing the ivr file, but it does not actually play anything, it simply
moves on to the next statement without letting the user enter in any digits.


*2. Executing the Dial command*

I can get the AGI to dial a number by using the command 'DIAL
SIP/<phone_number>" . But as soon as I try and pass any additional options,
it tells me there is no route to the host. I used to be able to run this
command : "DIAL SIP/27312048090 at cyb-voicetrading-a|30|L(6000000:61000:30000)",
but that gives me the following output :

AGI Script Executing Application: (DIAL) Options:
(SIP/27312048090 at cyb-voicetrading-a|30|L(6000000:61000:30000))
  == Using SIP RTP CoS mark 5
[Aug 13 14:31:13] WARNING[18947]: chan_sip.c:4089 create_addr: No such host:
cyb-voicetrading-a|30|L(6000000
[Aug 13 14:31:13] WARNING[18947]: app_dial.c:1444 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)


Any suggestions or comments would be greatly appreciated.

Ralfe

--
"He attacked everything in life with a mix of extraordinary genius and naive
incompetence, and it was often difficult to tell which was which."
- Douglas Adams
==================================

R a l f e P o i s s o n
ralfepoisson at gmail.com

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